| 00:00.27 | Juggie | dan2, disable printer port, serial port, and onboard soundcard |
| 00:01.53 | ZeroXeal | everytime I try to install asterisk no matter if its from the ports or a package (im on FreeBSD) I get this *** Error code 2 |
| 00:01.53 | ZeroXeal | Stop in /usr/ports/devel/pwlib. |
| 00:01.53 | ZeroXeal | *** Error code 1 |
| 00:02.08 | ZeroXeal | any one have any clue what the hell is up with that? |
| 00:02.30 | Qwell | try pasting more of the error to pastebin.com |
| 00:02.39 | Qwell | What you just pasted doesn't mean very much at all |
| 00:02.48 | ZeroXeal | alright |
| 00:02.50 | Mavvie | yes, what is the error before that? |
| 00:02.59 | Mavvie | (and OS version and friends too (uname -a) |
| 00:03.24 | StilexIP | is rtpproxy required when using ser and asterisk on the same box |
| 00:04.06 | ZeroXeal | Mavvie: where you speaking to me? |
| 00:04.20 | Mavvie | yes |
| 00:04.25 | ZeroXeal | FreeBSD 5.3 |
| 00:04.44 | ZeroXeal | gmake[2]: *** [/usr/ports/devel/pwlib/work/pwlib/lib/obj_FreeBSD_x86_r/qchannel.o] Error 1 |
| 00:04.44 | ZeroXeal | gmake[2]: Leaving directory `/usr/ports/devel/pwlib/work/pwlib/src/ptlib/unix' |
| 00:04.44 | ZeroXeal | gmake[1]: *** [opt] Error 2 |
| 00:04.44 | ZeroXeal | gmake[1]: Leaving directory `/usr/ports/devel/pwlib/work/pwlib' |
| 00:04.44 | ZeroXeal | gmake: *** [optnoshared] Error 2 |
| 00:04.44 | ZeroXeal | *** Error code 2 |
| 00:04.46 | ZeroXeal | Stop in /usr/ports/devel/pwlib. |
| 00:04.48 | ZeroXeal | *** Error code 1 |
| 00:04.50 | ZeroXeal | Stop in /usr/home/alex/asterisk. |
| 00:04.55 | ZeroXeal | sorry for the spam but that is the error |
| 00:05.09 | Mavvie | you missed again the most important line. |
| 00:05.12 | ZeroXeal | lol |
| 00:05.21 | Mavvie | first, go to pastebin.ca with your browser. |
| 00:05.25 | Qwell | Mavvie: The part where I said pastebin.com? :p |
| 00:05.26 | Mavvie | there, paste the 30 last lines. |
| 00:05.39 | Mavvie | Qwell: well, the two most important lines :-) |
| 00:05.42 | Qwell | ;] |
| 00:06.19 | ZeroXeal | I threw it on pastebin |
| 00:06.23 | ZeroXeal | zeroxeal |
| 00:06.25 | Qwell | url? |
| 00:06.36 | Qwell | we're lazy |
| 00:06.39 | ZeroXeal | http://pastebin.com/235550 |
| 00:06.40 | ZeroXeal | hehe |
| 00:06.56 | ZeroXeal | oh jesus |
| 00:06.58 | ZeroXeal | im a noob |
| 00:07.03 | ZeroXeal | it can't find a compiler |
| 00:07.10 | Schism | anyone have a working broadvoice patch for 1.0.5? |
| 00:07.14 | twisted | wheee |
| 00:07.17 | twisted | i love my laptop |
| 00:07.19 | twisted | dualing displays rock ;) |
| 00:07.22 | twisted | anywho. |
| 00:07.25 | ZeroXeal | hehe |
| 00:07.41 | Qwell | twisted: What laptop? |
| 00:07.43 | Schism | or know how to register w/ broadvoice unpatched? |
| 00:07.59 | Mavvie | ZeroXeal: that's cool, not many people get these. |
| 00:08.02 | twisted | Qwell, mine |
| 00:08.03 | twisted | :P |
| 00:08.04 | Qwell | Schism: I think the patch is to make them not pissed. I believe registering still "work" though |
| 00:08.05 | mikegrb | twisted: I have three displays so :p |
| 00:08.09 | Qwell | twisted: I mean what brand. :P |
| 00:08.18 | twisted | Qwell, dell |
| 00:08.26 | twisted | mikegrb, on your laptop? |
| 00:08.27 | ZeroXeal | Mavvie: do you know exactly what it is? lack of compiler? |
| 00:08.27 | Schism | I get registration errors trunking to them |
| 00:08.32 | Mavvie | ZeroXeal: I think that, if reproducable, mentionin it at on ports@freebsd.org is a good thing. |
| 00:08.34 | mikegrb | twisted: no :< |
| 00:08.37 | Qwell | yeah, co-worker has one of the newer Dell laptops, using dual display...looks nice on the LCD |
| 00:08.40 | twisted | mikegrb, ;) |
| 00:08.40 | mikegrb | twisted: desktop I'm on right now |
| 00:08.44 | *** join/#asterisk BoRiS (~boris@S01060040ca1e5b54.wp.shawcable.net) |
| 00:08.45 | Mavvie | /usr/include/c++/3.4/bits/locale_facets.h:2847: internal compiler error: Segmentation fault |
| 00:08.57 | mikegrb | working on one, irc on another and a full screen divx I'm watching with the wife on the third |
| 00:09.03 | Qwell | twisted: Do you know if jbot is linked to the karma on mantis, or if its seperate? |
| 00:09.09 | twisted | it's seperate. |
| 00:09.11 | ZeroXeal | this is a totally fresh install to |
| 00:09.14 | Qwell | ahh, ok |
| 00:09.25 | hermie | Qwell: jbot sits in a bunch of different channels |
| 00:09.31 | Qwell | hermie: yeah, I know |
| 00:09.41 | hermie | someday bugbot might do karma |
| 00:09.46 | *** join/#asterisk juice (~juice@mo-65-40-191-201.dyn.sprint-hsd.net) |
| 00:10.01 | Qwell | Who owns jbot? Just out of curiousity |
| 00:10.02 | Mavvie | ZeroXeal: there are no known problems with building it on the package building cluster. |
| 00:10.10 | hermie | Qwell: tim riker |
| 00:10.14 | Mavvie | (http://portsmon.firepipe.net/portoverview.py?category=&portname=pwlib&wildcard=) |
| 00:10.27 | Qwell | hermie: knew that too, heh. Who is he though? |
| 00:10.54 | ZeroXeal | Mavvie: this is strange |
| 00:11.21 | hermie | Qwell: the guy who made the tuxsreen |
| 00:11.25 | Qwell | ahh |
| 00:11.26 | hermie | tuxscreen |
| 00:11.30 | ZeroXeal | I might run some tests on the hardware this computer has given me problems before |
| 00:12.10 | ZeroXeal | arg today was a total waste |
| 00:14.14 | hermie | Qwell: problem is, there's no great way to parse karmas out of Mantis, other than to step through karma_view_transactions.php for each UID |
| 00:14.23 | Qwell | hermie: yeah |
| 00:15.31 | hermie | maybe I'll email malcolm about it, but I hate bugging him with stuff like that :) |
| 00:15.46 | Qwell | I wasn't suggesting it be done or anything, heh |
| 00:15.52 | Qwell | was just curious if it was linked at all |
| 00:16.23 | Qwell | but, erm, it would be stored in a DB, no? |
| 00:16.25 | hermie | well, other people have asked about it for bugbot (who sits on the -bugs and -dev channel) and I've kinda forgot about it |
| 00:16.35 | hermie | Qwell: yeah, Digium's db |
| 00:16.53 | Qwell | ahh, external bot |
| 00:19.19 | *** join/#asterisk VoipLugNut (~stuart@243-73.8-67.tampabay.rr.com) |
| 00:19.26 | Qwell | ahh, hermie...you submitted that patch for me yesterday :p |
| 00:20.26 | hermie | the one with the ' ' instead of a /? |
| 00:20.30 | Qwell | yeah, heh |
| 00:20.59 | StilexIP | is anyone currently using ser along with asterisk?? |
| 00:21.46 | hermie | StilexIP lots of people are |
| 00:21.49 | Qwell | hermie: yeah, I wasn't doing very good at making the patch. never really used cvs before. I looked at the command in your diff, and it was weird. diff -u -u -r 1.4 Makefile, or some such |
| 00:22.05 | Qwell | I assume that came out of cvs? |
| 00:22.06 | hermie | Qwell: I saw you just edited my patch |
| 00:22.12 | Qwell | I did |
| 00:22.19 | hermie | Qwell: you just use 'cvs diff -u <file>' |
| 00:22.24 | hermie | times were the same :) |
| 00:22.29 | Qwell | yeah, heh |
| 00:23.21 | Qwell | I'll have to remember how to do it from cvs for the next time |
| 00:23.37 | StilexIP | hermie: do you know how to setup the ser.cfg correctly to make it work with rtpproxy or do we even need to use rtpproxy / because ser and * are on the same box |
| 00:23.54 | hermie | Qwell: you can always get help with stuff like that on #asterisk-bugs |
| 00:24.16 | Qwell | ahh, didn't realize that channel existed |
| 00:24.25 | hermie | StilexIP: I don't use SER... lots of other people are tho |
| 00:24.42 | hermie | Qwell: shhh.. don't let too many people know :) |
| 00:24.46 | Qwell | wow, there are quite a bit, hmm? |
| 00:25.04 | hermie | of what? |
| 00:25.14 | Qwell | #asterisk-* channels |
| 00:25.23 | Qwell | I just whois'd you, heh |
| 00:25.40 | hermie | i'm not even in _all_ of them |
| 00:26.26 | Chuji | He's not in #asstricks |
| 00:27.27 | buddah | exten => 2457,102,VoiceMail,b2457 |
| 00:27.33 | buddah | is that how to get it to go to vmail if busy? |
| 00:28.52 | Qwell | its VoiceMail(b2457), isn't it? |
| 00:29.48 | *** join/#asterisk MrEntropy (~entropy@ppp136-158.lns1.adl2.internode.on.net) |
| 00:30.27 | hermie | MrEntropy is always popping in at random times :) |
| 00:30.45 | MrEntropy | hermie: you don't say =) |
| 00:30.52 | buddah | exten => 2457,2,VoiceMail,u2457 |
| 00:30.55 | buddah | i got that and it works |
| 00:31.00 | buddah | when the line is empty |
| 00:31.05 | buddah | but if its busy, it doesnt work |
| 00:31.14 | Qwell | Should it be comma, or ()? |
| 00:31.15 | MrEntropy | hermie: i like to keep people on their toes |
| 00:31.26 | hermie | buddah, pastebin your dialplan |
| 00:31.29 | buddah | the doc i was looking at said , |
| 00:31.30 | buddah | ok |
| 00:31.31 | buddah | hold on |
| 00:31.44 | Qwell | exten => s-BUSY,1,Voicemail(b${ARG1}) ; straight from the demo |
| 00:31.44 | hermie | MrEntropy: you're so disorderly! |
| 00:32.18 | buddah | http://pastebin.ca/5004 |
| 00:32.23 | MrEntropy | hermie: definately, just enough to keep you guessing =) |
| 00:33.12 | buddah | so if i add the (2457) it wont prompt for the box # when i dial in for mail |
| 00:35.50 | *** join/#asterisk wolfson (~hehe@bcp-68-187-181-159.man.nc.charter.com) |
| 00:38.39 | *** join/#asterisk JamesDotCom (~james@sweep.bur.st) |
| 00:39.37 | chrisfrog | right, time to give up with installing the asterisk webmin mod. as i am getting no where |
| 00:40.08 | hermie | buddah: you aren't getting voicemail when an exten is busy? |
| 00:40.50 | buddah | correct |
| 00:40.51 | StilexIP | would i need to use rtpproxy if my ser and asterisk are on the same box |
| 00:40.52 | buddah | just getting nothing |
| 00:41.28 | hermie | buddah: nothing at all? |
| 00:41.31 | buddah | nope |
| 00:42.24 | freat | buddah: do you have the mailbox defined in voicemail.conf ? |
| 00:42.49 | buddah | yeah |
| 00:42.56 | buddah | it works when nobody is on the phone |
| 00:43.02 | freat | oh heh checking out your pastebin now... |
| 00:43.02 | hermie | buddah: try using the DIALSTATUS like they do in the standard extension macro |
| 00:43.15 | hermie | (in fact, you might just wanna use the macro...) |
| 00:43.17 | buddah | what is that? |
| 00:43.39 | hermie | buddah look in the sample extensions.com |
| 00:44.10 | buddah | hmmm actually is it because call waiting |
| 00:44.30 | buddah | the phone beeps |
| 00:44.33 | tangel | how can i pick up on my fxo port by dialing an extension? |
| 00:44.44 | tangel | i want to be able to pick up my house phone but i don't want * to always pick it up |
| 00:44.52 | tangel | (so the other straight analog phones ring as normal) |
| 00:45.37 | buddah | when i dial to the phone while its on another call |
| 00:45.41 | buddah | i dont even here it dialing |
| 00:45.53 | buddah | i'd think i'd at least hear that |
| 00:47.28 | beto75 | hello guys , I hear a HORRIBLE noise like Bang, bang, bang with a Snom 200 phone (g711u) |
| 00:47.44 | beto75 | only when a voice finishes |
| 00:48.08 | mikegrb | well the noise isn't because the voice is finishing |
| 00:48.14 | mikegrb | the voice is finishing because of the noise |
| 00:48.17 | tzanger | ~seen kram |
| 00:48.18 | jbot | kram <~mark@kram.digium.sponsor.pdpc> was last seen on IRC in channel #asterisk, 5h 34m 24s ago, saying: 'also i think the bug guidelines themselves need some clarification'. |
| 00:48.25 | mikegrb | see, the person you were talking to is a mob boss |
| 00:48.30 | beto75 | no Mike |
| 00:48.34 | mikegrb | the oposing mob family shot him |
| 00:48.41 | mikegrb | that is the bang bang noise you were hearing |
| 00:48.43 | *** join/#asterisk Delmar (~Delmar@222-152-57-78.adsl.inspire.net.nz) |
| 00:48.50 | mikegrb | then his voice stopped because he was dead |
| 00:49.00 | Dagrim | hahaha |
| 00:49.06 | beto75 | for example if you call me ,, when you finish HELLO I hear BANG BANG, then you say other word and when finish the complete work BANG BANG |
| 00:49.08 | mikegrb | the moral of the story is don't talk to mob bosses and you won't hear bang bang bang through your snom phone |
| 00:49.22 | buddah | ok it works |
| 00:49.23 | *** join/#asterisk marocon-tim (~tim@207.237.145.64) |
| 00:49.25 | mikegrb | maybe the phone is a fan of chitty chitty bang bang |
| 00:49.25 | mikegrb | ? |
| 00:49.27 | beto75 | while the words are trough no problem ,, just when you finish talking |
| 00:49.28 | Dagrim | mikegrb: especially on a Snom phone right? |
| 00:49.34 | Dagrim | lol |
| 00:49.41 | buddah | just no ring, just silence, then 40 sec and then busy voicemail |
| 00:49.47 | Delmar | lookin for some suggestions for dealing with Echo using X100P cards (other than the chan_zap.c read_size modification). can anyone help? |
| 00:49.58 | marocon-tim | Anyone notice recently that AGIs can't do post call processing anymore? Mine are getting killed immediately after I read the dial status after a hangup. |
| 00:50.00 | mikegrb | beto75: you could tell them to hum while they are not talking |
| 00:50.14 | twisted | or stop slapping the table when they're done talking |
| 00:50.15 | Dagrim | lol |
| 00:50.17 | mikegrb | beto75: or you could make sure silence detection/supression is off |
| 00:50.18 | Qwell | mikegrb: maybe he could hum |
| 00:50.32 | hermie | buddah: what version |
| 00:50.56 | mikegrb | Qwell: It can get boring at times, got to make sure you and Dagrim don't fall asleep |
| 00:51.07 | beto75 | Mike , I thnk that will be OFF from asterisk my 2 end points (Welltech gateway and snom phone) has VAD off |
| 00:51.10 | Dagrim | beto75: yea def. check the silence suppression like mike said |
| 00:51.36 | Dagrim | mikegrb: sleep? whats that? =p |
| 00:51.39 | srt | i have a problem with the way chan_iax2 chooses its source ip address when respondin |
| 00:51.42 | srt | g to incoming requests, can anybody help me where to start looking? |
| 00:51.50 | Dagrim | mikegrb: Oh you mean wait() .. hah hah.. just kidding. |
| 00:51.58 | mikegrb | Dagrim: ;) |
| 00:52.09 | Dagrim | zzZZZzzzZzZzz() |
| 00:52.18 | srt | i.e. it doesnt use the destination address of the packet received as the source address for the response... |
| 00:52.57 | mikegrb | beto75: that's the main thing I would check if it doesn't help I would try sending a message to the users mailing list, make sure you tell them VAD and such is already off |
| 00:53.13 | Delmar | srt, are u meaning u have more than one interface in your box and the reply is sent "from" the wrong IP kinda thing? |
| 00:53.15 | mikegrb | beto75: very very many people on mailing list, maybe someone has seen it before |
| 00:53.26 | mikegrb | beto75: oh, don't foirget to search the list archives first |
| 00:53.27 | Qwell | Whats vad stand for? I mean, I know vad=silence suppression, but... |
| 00:53.51 | srt | Delmar: yes. its eth0 and eth0:1... |
| 00:53.53 | Delmar | srt, use the bindaddr= in iax.conf. |
| 00:54.00 | Dagrim | Qwell: i know that if mine is set to anything but false my quality is HORRID |
| 00:54.17 | Qwell | Dagrim: Is that what your problem ended up being the other night? |
| 00:54.24 | srt | yes that works but then i cant handle connections on other interfaces |
| 00:54.27 | mikegrb | Qwell: very anal detection? |
| 00:54.37 | srt | i also have a tun0 for vpn that should be handled, too |
| 00:54.48 | Qwell | mikegrb: I was thinking somewhere more along the lines of variable audio detection, but that works too. :p |
| 00:54.54 | Delmar | srt, i see. now you are getting tricky :P |
| 00:54.57 | Dagrim | Qwell: combination of that.. medioker bandwith.. and Having my menus split up into way to many seperate samples.. |
| 00:55.02 | mikegrb | Qwell: my bet would be on your guess |
| 00:55.11 | Qwell | mikegrb: is that right? |
| 00:55.15 | Delmar | srt, does it choose the same IP address for all replies? |
| 00:55.15 | srt | yes asterisk always takes the ip of eth0:1 so connecting with that ip works |
| 00:55.19 | Qwell | I just kinda made that up |
| 00:55.25 | mikegrb | Qwell: I wouldn't be suprised if it was ;) |
| 00:55.26 | srt | but i think thats rather a workaround ;) |
| 00:55.55 | Qwell | mikegrb: Voice activity detection |
| 00:56.04 | mikegrb | ! |
| 00:56.05 | mikegrb | ja |
| 00:56.08 | Dagrim | Qwell: i guess it was using silence suppression on the menu samples. lol.. which are in gsm, so oh yea thats gonna work REAL well lemme tell you.. lol |
| 00:56.10 | Qwell | makes more sense, heh |
| 00:56.13 | srt | connection on an interface with only one ip (e.g. my vpn interface) works |
| 00:56.25 | *** part/#asterisk marocon-tim (~tim@207.237.145.64) |
| 00:56.36 | Qwell | Dagrim: so you were getting distorted sound after every sample, or something? |
| 00:57.01 | Dagrim | Qwell: what is your exact line for vad? Does it seriously look like: vad=silence suppression theres a space in there? hmmmm |
| 00:57.16 | Qwell | Dagrim: no, that was my interpretation of the two, being the same |
| 00:57.28 | Qwell | in other words, "vad is the same thing as silence suppression" |
| 00:57.28 | Delmar | Does anyone have any ideas for dealing with Echo Cancellation? Calls between SIP phones/softphones etc are all working great, but anything via the X100P's have massive echo. Have tried messing with tx/rx gain. am about to apply READ_SIZE mod to chan_zap.c but other than this im at a loss ... anyone got any other ideas? |
| 00:57.35 | Dagrim | Qwell: it was clipping the front and end of each sample.. PLUS like differents parts of my menus have different samples.. etc.. one sample could be anywhere from 1 to 5 words on the menu |
| 00:57.53 | Dagrim | Qwell: so vad=true turns S suppression on.. gotcha |
| 00:57.54 | Qwell | ahh, yeah, that would make sense that it was distorted then |
| 00:57.58 | Qwell | Dagrim: no, heh |
| 00:58.00 | *** join/#asterisk shell (shell@200.56.130.231) |
| 00:58.18 | Qwell | Dagrim: I don't think the config option even has "vad" in it |
| 00:58.20 | srt | Delmar: do you have yet another idea or should i submit a bug report on that one? |
| 00:58.24 | mikegrb | Qwell: just found out my mother in law is buying us a new car seat |
| 00:58.27 | Dagrim | hmm.. |
| 00:58.36 | Dagrim | Qwell: mine did.. lol |
| 00:58.36 | Qwell | mikegrb: erm? |
| 00:58.43 | mikegrb | Qwell: an $180 leather eddie bauer car seat! |
| 00:58.46 | Qwell | Dagrim: oh, I don't know anything about it |
| 00:58.51 | Dagrim | right on... |
| 00:58.52 | mikegrb | Qwell: it has frik'n cup holders! |
| 00:58.53 | Qwell | mikegrb: child seat? |
| 00:58.56 | mikegrb | yes |
| 00:59.00 | Dagrim | well that was my prob if anybody ever runs into that again =) |
| 00:59.03 | Qwell | heh, saw those at babies'r'us |
| 00:59.14 | mikegrb | $180! leather! cup holders! damn! |
| 00:59.16 | Qwell | graco for me, thanks :p |
| 00:59.29 | mikegrb | but hey, if she wants to spend the money, I'll enjoy using it |
| 00:59.33 | Qwell | mikegrb: how old is (s)he? |
| 00:59.35 | Dagrim | mikegrb: nice .. lucky kid.. |
| 00:59.40 | *** part/#asterisk shell (shell@200.56.130.231) |
| 00:59.40 | mikegrb | I will enjoy selling it on ebay even more though ;) |
| 00:59.46 | mikegrb | he is 4 months |
| 00:59.49 | Qwell | ahh |
| 01:00.01 | Qwell | mine should be popping out within a few days/weeks now, heh |
| 01:00.12 | mikegrb | heh |
| 01:00.26 | Qwell | yesterday was "full term" |
| 01:00.27 | mikegrb | mine was born with flash lights in a 90 degree room |
| 01:00.32 | Qwell | nice |
| 01:00.34 | Dagrim | brb |
| 01:00.37 | mikegrb | and two months early |
| 01:00.40 | mikegrb | but he is big now |
| 01:00.41 | Qwell | ouch |
| 01:01.08 | mikegrb | was born just after huricane ivan |
| 01:01.22 | Qwell | which would explain the flashlights...right |
| 01:02.01 | mikegrb | yes |
| 01:02.11 | mikegrb | and the 90 degree operating room :/ |
| 01:02.26 | mikegrb | they had a box fan with a long extention cord pointing on my wife |
| 01:02.32 | Qwell | it was actually at a hospital, with no power? |
| 01:03.16 | Dagrim | wow.. that sounds insane |
| 01:03.39 | mikegrb | Qwell: yes |
| 01:03.45 | Qwell | hmm |
| 01:03.58 | mikegrb | we had power for lights and tv and stuff then moved to labor and delivery operating room |
| 01:04.15 | mikegrb | the smart engineer who designed the place decided it didn't need any power from generator in there |
| 01:04.18 | Delmar | bah. all going well you dont need power... some light, and a bottle of air to kick start his/her lungs just in case... |
| 01:05.01 | Delmar | my girl was born last 16th. |
| 01:05.26 | mikegrb | Delmar: this was 2 /months/ before concidered full term |
| 01:06.01 | Delmar | 32 weeks is a little early alright. |
| 01:06.06 | *** join/#asterisk MrEntropy (~entropy@ppp136-158.lns1.adl2.internode.on.net) |
| 01:06.10 | Qwell | thats more like 29, no? |
| 01:06.29 | mikegrb | it was right at 30 IIRC |
| 01:06.34 | *** join/#asterisk batok (~director@200.56.164.1) |
| 01:06.45 | Qwell | My wife tried to explain to me how weeks and months work together, but I didn't believe it. :p |
| 01:06.55 | Qwell | I stuck to my story that 13 weeks == 3 months |
| 01:07.04 | mikegrb | Qwell: :D |
| 01:07.10 | Delmar | nah 40 weeks is full term. |
| 01:07.10 | buddah | Jan 30 17:06:12 NOTICE[6612]: app_dial.c:884 dial_exec_full: Unable to create channel of type 'SIP' (cause 3) |
| 01:07.13 | Qwell | I guess with this, it somehow defies the calendar :P |
| 01:07.14 | buddah | anyone know what that is? |
| 01:07.19 | K-Sensei | I'm having a problem with the latest stable CVS release (v1-0)... I have two extensions set up with IAX2, and I am using IaxPhone on two computers trying to call one from the other. Whenever I do, asterisk bombs out with a "Floating point exception". I had a feeling it might be music on hold... so I set up an extension via "exten => 6601,1,WaitMusicOnHold(30)". Whenever I call it, I get this: |
| 01:07.30 | K-Sensei | <PROTECTED> |
| 01:07.30 | K-Sensei | <PROTECTED> |
| 01:07.30 | K-Sensei | Floating point exception |
| 01:07.31 | Qwell | Delmar: see, I'm being told 37 |
| 01:07.48 | Delmar | K-Sensei heh i think i know what that is. :P |
| 01:07.52 | tzanger | K-Sensei: what CPU is this on? |
| 01:08.00 | K-Sensei | Intel Xeon |
| 01:08.33 | K-Sensei | I'm running an SMP kernel also. |
| 01:08.33 | tzanger | hmm |
| 01:08.35 | K-Sensei | So, what is it? |
| 01:08.36 | tzanger | typically you get those problems if it's been compiled for a different machine |
| 01:08.47 | K-Sensei | I compiled it on here from the sources in CVS. |
| 01:08.57 | tzanger | I said typically. :-) |
| 01:08.59 | Delmar | K-Sensei ok thats not quite the same actually..... problem i was having would occur when call ended... |
| 01:09.05 | *** join/#asterisk pepem_sk (~pepem_sk@adsl-data-114.84-47-111.telecom.sk) |
| 01:09.12 | K-Sensei | What did you do to fix it? |
| 01:09.27 | K-Sensei | I thought it might be mpg123 also, so I uninstalled it and I installed asterisk-addons and I'm using format_mp3 now instead... but I still get the same error. |
| 01:09.37 | Qwell | K-Sensei: run it in gdb, see if you can get a backtrace |
| 01:09.51 | Delmar | well its not the same problem but.. i removed the rate_engine.conf |
| 01:09.51 | K-Sensei | Any good crash course on gdb I could look at? |
| 01:09.52 | K-Sensei | heh |
| 01:10.14 | Qwell | K-Sensei: not sure. I bet if you look at the asterisk-users archive, you'll see bkw_ explaining it a few hundred times though, heh |
| 01:10.20 | Delmar | i removed rate_engine.conf from /etc/asterisk .. and everything worked. probably didnt like the default and non-working config in there. |
| 01:10.26 | greg_work | anyone know if theres a debian package for tiff2ps/ps2pdf ? |
| 01:10.50 | Qwell | greg_work: tried an apt search for ps? |
| 01:10.57 | *** join/#asterisk nextime (~nextime@ns0.nexlab.net) |
| 01:11.04 | greg_work | yeah but theres nothing specific |
| 01:11.38 | Delmar | pdfjam ? |
| 01:12.29 | K-Sensei | okay thanks Qwell, I'll look into that. |
| 01:12.36 | K-Sensei | I'll also try your idea, Delmar. |
| 01:12.46 | Delmar | im not sure if thats gonna be useful... |
| 01:12.54 | Delmar | but i see it there... pdf utilities... sounds worth a look. |
| 01:13.29 | greg_work | ah, apt-file (which i didnt know about till now) tells me it's in gs-common |
| 01:13.55 | K-Sensei | Delmar: I have no "rate_engine.conf" file in /etc/asterisk. |
| 01:14.11 | drumkilla | K-Sensei: http://dev.asteriskdocs.org/index.php/Gdb |
| 01:14.19 | Delmar | there u go :P |
| 01:14.21 | K-Sensei | drumkilla: Thanks, man. |
| 01:14.25 | drumkilla | no problem ... |
| 01:14.28 | drumkilla | it's pretty short, though ... |
| 01:14.55 | K-Sensei | allright, I gtg shopping (wife is bugging the hell out of me) |
| 01:14.58 | K-Sensei | be back in a few |
| 01:15.02 | Qwell | drumkilla: :) |
| 01:15.28 | Qwell | hmm |
| 01:15.30 | Qwell | ~gdb |
| 01:15.31 | jbot | methinks gdb is The GNU Debugger. URL: http://www.gnu.org/software/gdb/ or http://sources.redhat.com/gdb/ |
| 01:15.36 | drumkilla | greg_work: I think those come with libtiff |
| 01:16.06 | greg_work | drumkilla: yeah, just found both, libtiff-tools and gs-common |
| 01:16.12 | drumkilla | cool |
| 01:18.30 | Dagrim | Anybody know anything about Front side buses and Frequency select jumpers? lol |
| 01:18.49 | Dagrim | I have this awesome box here .. but I dont think its all set right.. blah |
| 01:19.08 | Qwell | Dagrim: I always google the manual. They change way too much from board to board |
| 01:19.14 | Dagrim | I know.. |
| 01:19.22 | Qwell | and from cpu to cpu, heh |
| 01:19.26 | Dagrim | Qwell: it is 2 diff things tho right? FSB and Freq selection? |
| 01:19.32 | Qwell | usually |
| 01:19.45 | Qwell | always? dunno |
| 01:19.48 | Dagrim | Okay.. awesome.. then there MUST be other jumpers on there.. |
| 01:19.54 | Dagrim | =) thanks qwell |
| 01:20.00 | Qwell | sometimes there aren't FSB jumpers |
| 01:20.17 | Qwell | my board has neither, its all in the bios |
| 01:21.11 | greg_work | in zapata.conf, what is the faxdetect option for? I mean, i understand faxdetect=incoming, but what does faxdetect=outgoing do? |
| 01:21.33 | Qwell | probably so it doesn't mangle the outgoing call somehow |
| 01:21.37 | Qwell | Don't listen to me though |
| 01:22.20 | mikegrb | I always listen to Qwell |
| 01:23.52 | drumkilla | I got another offer to pick up 10 million dollars ... |
| 01:23.56 | drumkilla | everyone just wants to give me money |
| 01:24.04 | greg_work | the only thing i can think of is if you're on an outgoing call and someone on the remote end decides to send a fax at you, and you gave a fax extension in the current context, it'll receive it .. but how that is usful is beyond me ;p |
| 01:24.18 | hermie | drumkilla: WAS IT FROM MR. QZ. MADELA? |
| 01:24.28 | drumkilla | HOW'D YOU KNOW! |
| 01:24.51 | hermie | HE CONTACTED ME ABOUT A HIGHLY SENSITIVE MATTER |
| 01:24.58 | drumkilla | no way! |
| 01:25.00 | greg_work | hermie: i can foward you some other great m|or|tgage offers if you want. i can get some great prices on cilias and rolex watches, too |
| 01:25.04 | drumkilla | don't do it! |
| 01:25.26 | hermie | http://www.j-walk.com/other/conf/ |
| 01:26.40 | Nugget | "Unlike the elementary version often seen, GNU Hello processes its argument list to modify its behavior, supports greetings in many languages, and includes a mail reader" |
| 01:27.56 | dan2 | twisted: ping |
| 01:28.50 | greg_work | OMG, someone misspelled our company name in our fax machine's header! |
| 01:29.00 | Nugget | heh |
| 01:29.58 | Qwell | If your company name is simple, smack them |
| 01:30.12 | greg_work | oh, no they didn't.. the phone company misspelled it in our caller id ! |
| 01:30.18 | buddah | will phones not registered with sip work with a registered g729 channel for voicemail? |
| 01:30.18 | Qwell | oh, lovely |
| 01:30.55 | greg_work | hm, only for that line, apparently |
| 01:31.23 | Qwell | some kid making $10/h was typing them in manually on each line, heh |
| 01:31.57 | greg_work | they put "mc" instead of "mac" |
| 01:32.05 | Qwell | thats forgivable |
| 01:32.06 | greg_work | so its not a HUGE deal, but still.. |
| 01:32.16 | drumkilla | you're a sub-franchise of mcdonalds! |
| 01:32.23 | greg_work | macdonalds? :) |
| 01:32.34 | Qwell | There is a MacDonalds in like ireland or something |
| 01:32.39 | Qwell | mcdonalds tried to sue them |
| 01:32.44 | greg_work | i wonder if you'd get sued :) hehheh |
| 01:32.47 | Qwell | They've had their name for like 100 years, heh |
| 01:32.53 | greg_work | actually, we have a supplier caled "MacDonald and Sons" |
| 01:33.08 | Qwell | as long as it isn't a place to buy food, its ok |
| 01:33.15 | greg_work | but they sell plumbing parts, not hamburgers |
| 01:33.18 | Delmar | Is anyone here familiar with X100's ? |
| 01:33.21 | Qwell | MacDonald and Sons grocer however, would probably get sued, heh |
| 01:33.57 | greg_work | hm, so now i'm not sure what i want to do with faxes |
| 01:34.10 | Qwell | greg_work: fax tag |
| 01:34.11 | greg_work | we have a dedicated fax line and a machine right now |
| 01:34.16 | Delmar | greg_work ooo u are playing with faxes and asterisk? |
| 01:34.33 | greg_work | but the line is also connected to our phone system so it can be used for outgoing calls |
| 01:35.01 | greg_work | at the same time, i've wanted for a while (adn just haven't had time) to get a software-based fax system going (hylafax) |
| 01:35.07 | greg_work | but now that * can receive them ... |
| 01:35.28 | greg_work | (this comes in on POTS to a TDM400P btw) |
| 01:35.30 | `Sauron | greg: It's relatively easy |
| 01:35.33 | Delmar | thats something i wanna do later.. have no idea what u can do but.. kinda wondering if Asterisk can act as like.. a fax switch, and route the detected call to a defined extension... which would be a PC running relayfax or some crap... |
| 01:36.02 | greg_work | should I have that line (Zap/4-1) go directly to the fax (connected to spa-2000), or receive with rxfax() ? |
| 01:36.13 | greg_work | Delmar: yes, exactly |
| 01:36.19 | Delmar | but i dont see why Asterisk couldnt detect the call, then pipe the call into some fax software on the box itself.. rather than routing it out an extension, analog again.. then to a fax device... |
| 01:36.26 | greg_work | Delmar: when it detects a fax tone, it jumps to fax,1 |
| 01:37.18 | Delmar | someone will build some cool little faxdriver.so or something and Asterisk will start to take care of fax reception and stuff.. im sure :P |
| 01:37.22 | Dagrim | Yea.. Isnt there a way to just have it write them as a image file or something? |
| 01:37.37 | Dagrim | Thatd be awesome. |
| 01:37.43 | greg_work | Delmar: exten => fax,1,Macro(fax) [macro-fax] exten => s,1,SetVar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif) exten => s,2,SetVar(EMAILADDR=${FAX_RX_EMAIL}) exten => s,3,rxfax(${FAXFILE}) |
| 01:37.49 | Delmar | i dont think Asterisk knows how to speak.. fax talk.. does it? its just going to route the "audio". |
| 01:37.58 | Dagrim | I dont see why people use fax machines much anymore anyways.. |
| 01:38.05 | Delmar | greg_work what was all that haha |
| 01:38.05 | Qwell | greg_work: It isn't that easy, is it? |
| 01:38.18 | greg_work | i just set it up, barely had to do anything |
| 01:38.23 | Qwell | heh |
| 01:38.29 | Delmar | what exactly is that doing tho? |
| 01:38.32 | Qwell | and it works good, with those those couple lines? |
| 01:38.33 | greg_work | i'm going to write a web-based fax inbox though |
| 01:38.34 | *** part/#asterisk Brixius (Brixius@c-24-118-4-197.mn.client2.attbi.com) |
| 01:38.43 | Delmar | greg_work that would rock. |
| 01:38.47 | Qwell | Delmar: looks like its sending a tif file as a mail to somebow |
| 01:38.59 | Qwell | I don't know what rxfax does though, of course... |
| 01:39.04 | Dagrim | greg_work: So you where is the macro part? Built into * or no? |
| 01:39.10 | Delmar | greg_work, is that making Asterisk receive the fax ... dumping it to a .tif ? |
| 01:39.14 | greg_work | rxfax is a * function |
| 01:39.19 | Dagrim | wow. |
| 01:39.19 | greg_work | it puts it into a tiff file |
| 01:39.24 | Delmar | fuck me. |
| 01:39.26 | Delmar | thats wicked. |
| 01:39.28 | greg_work | i'm using 1.0.5 stable |
| 01:39.36 | Qwell | thats pretty damn simple |
| 01:39.36 | *** join/#asterisk niZon (ilt@S0106deadbeef6977.wp.shawcable.net) |
| 01:39.42 | greg_work | from what i can tell, txfax() is not built in (yet?) |
| 01:39.42 | Dagrim | yup |
| 01:40.04 | Dagrim | Hmmm.. of course.. Rxfax probably just came out then? |
| 01:40.10 | greg_work | i had some ideas from playing with hylafax of what i wanted to be able to do |
| 01:40.14 | greg_work | i dunno |
| 01:40.42 | greg_work | for example - the fax inbox would basically be like a web-based email client, except shared between everyone |
| 01:40.50 | greg_work | when you click on a fax, it downloads a pdf.. |
| 01:40.55 | Delmar | imagine if you could loginto your voicemail, manage your fax queue, and have Asterisk forward the Fax to a number and stuff like that.. but... fuck that.. a web-based fax-inbox to look at them is enuf... |
| 01:41.07 | Delmar | we gotta get people away from this old primative Fax nonsense :P |
| 01:41.17 | greg_work | then you can add advanced features - filtering to a given folder based on callerid |
| 01:41.30 | Delmar | yep. |
| 01:41.35 | Delmar | get to work :P |
| 01:41.41 | Qwell | Then you just need some OCR capabilities, to grep out a "To:" field, heh |
| 01:41.41 | Dagrim | hmm nice |
| 01:41.46 | greg_work | or go even further, do OCR on it and do some regex matches for "attention: greg" and move to a "greg" folder, as well as email that user saying they have a new fax |
| 01:41.57 | Delmar | yup. |
| 01:42.00 | Delmar | sounds wicked dude. |
| 01:42.10 | Delmar | i expect delivery later tomorrow :P |
| 01:42.18 | greg_work | you may just get it ;) |
| 01:42.30 | Delmar | lol no way. |
| 01:42.56 | `Sauron | Hum Di Dum. |
| 01:43.13 | greg_work | hm, not sure where it stores information on it though |
| 01:43.13 | Delmar | im just gonna knock this pain in the arse echo problem out of the park then i might play with that rxfax thingie. |
| 01:43.15 | Qwell | oh, then...get something like festival to call and speak the fax to Greg |
| 01:43.25 | `Sauron | Does anyone know if * can talk with plain modems, to switch data calls? |
| 01:43.35 | `Sauron | Alternately, I guess I could get a T1 card, eh. |
| 01:43.48 | greg_work | `Sauron: you can't detect data calls really |
| 01:44.04 | `Sauron | You mean data vs. voice? |
| 01:44.06 | greg_work | fax machines make noise when they're calling out, modems don't make noise till they hear the remote end do it |
| 01:44.16 | greg_work | yes, if that's what you're talking about ;) |
| 01:44.21 | `Sauron | or you mean fax vs. data? |
| 01:44.25 | `Sauron | Well |
| 01:44.37 | `Sauron | Here's the scenario, and it sort of ties into what you're talking about |
| 01:44.40 | Delmar | nah... u can make the modem generate a CNG tone in the dial string.... |
| 01:44.53 | greg_work | you can detect fax vs data, the same way you detect fax vs voice -- but you can't do voice vs data, or voice vs fax vs data |
| 01:44.59 | `Sauron | At work, we have a box, with some old Digi PRI cards in it |
| 01:45.21 | fearnor | define what exactly you mean "data" |
| 01:45.23 | greg_work | Delmar: ok even if thats true, still limited use. you'd have to know you have to do it |
| 01:45.25 | `Sauron | In linux, the channels on the PRI ports, show up as 48 modems |
| 01:45.31 | *** join/#asterisk Guest^DJ (~some@211.24.146.11) |
| 01:45.39 | Guest^DJ | hi guys |
| 01:45.41 | *** join/#asterisk UncleBill (WildX@c-67-161-7-70.client.comcast.net) |
| 01:45.42 | fearnor | data as in modem? you *conceivably* can distinguish it from fax. |
| 01:45.46 | UncleBill | hello friends |
| 01:45.48 | `Sauron | Now, the code I wrote to handle fax processing into email, is old and crufty |
| 01:46.00 | `Sauron | However, the machine also does modem dialups |
| 01:46.03 | fearnor | data as in isdn clear-channel call? conceivably, libpri should give you the calltype, I think. |
| 01:46.08 | greg_work | `Sauron: have you ever looked at hylafax? |
| 01:46.16 | `Sauron | yeah |
| 01:46.19 | `Sauron | it sucks ass |
| 01:46.26 | greg_work | oh? |
| 01:46.28 | Nugget | hylafax is a bloated, corpulent mess. |
| 01:46.30 | `Sauron | At least it did 3 years ago when I wrote this. |
| 01:46.45 | `Sauron | It was much better to use mgetty, and write 500 lines of perl code to do the rest |
| 01:46.56 | `Sauron | Well, approx. 500 lines |
| 01:47.26 | Delmar | grrrr. |
| 01:47.31 | greg_work | ah crap. is rxfax an addon? |
| 01:47.34 | Delmar | this X100 shit is driving me up the wall. |
| 01:48.09 | `Sauron | 399 lines of code, total |
| 01:48.16 | `Sauron | [dominic@aus-dialup bin]$ wc -l process_incoming_fax.pl /etc/mgetty+sendfax/new_fax |
| 01:48.17 | `Sauron | <PROTECTED> |
| 01:48.17 | `Sauron | <PROTECTED> |
| 01:48.17 | `Sauron | <PROTECTED> |
| 01:49.06 | `Sauron | Hum. |
| 01:49.52 | Delmar | damn it. when an incomming call comes in on the X100P, most of the time * decides that the line has gone into a strange state and all kinds of crap. i dont think line conditions are being detected properly and stuff. |
| 01:50.16 | `Sauron | Hum. |
| 01:50.25 | hermie | Delmar: genuine X100P (TM) ? |
| 01:50.29 | Delmar | not only that but, even changing the READ_SIZE in chan_zap.c I still have massive echo. |
| 01:50.48 | StilexIP | http://pastebin.ca/5006 |
| 01:50.56 | `Sauron | Greg, if you want to talk about faxing (inbound and outbound), feel free to get with me at some point. I gotta finish converting my dialplan to sql |
| 01:50.59 | `Sauron | :) |
| 01:51.01 | StilexIP | can someone take a look at that, trying to set up ser + asterisk |
| 01:51.03 | Delmar | its the OEM from Digitnetworks, and they say time and time again, that the card is exactly the same. |
| 01:51.13 | fearnor | delmar: it isn't. |
| 01:51.19 | Delmar | how so? |
| 01:51.26 | fearnor | jesus |
| 01:51.34 | fearnor | BE-FUCKING-CAUSE |
| 01:51.42 | fearnor | just because the chipset is the same and pci ids are the same |
| 01:51.52 | fearnor | it doesn't mean all the *components* on the PCB are the same |
| 01:51.52 | greg_work | `Sauron: enjoy :p i'm working on this now |
| 01:52.00 | *** join/#asterisk okieplaya (~jjj@ip68-229-252-53.ok.ok.cox.net) |
| 01:52.14 | greg_work | `Sauron: just downloading asterisk-addons .. need to find the source to rxfax() so i can see what it's doing |
| 01:52.14 | Chuji | ~x100p |
| 01:52.15 | jbot | methinks x100p is an obsolete card, copied by far too many people |
| 01:52.19 | `Sauron | You can call me if you want. I need to test my FWD iax2 connection |
| 01:52.22 | `Sauron | ;) |
| 01:52.40 | fearnor | regardless, even original x100p is shit :P |
| 01:52.48 | greg_work | hm, i wish mine even worked. is fwd working yet? |
| 01:52.52 | fearnor | clones are worse |
| 01:52.59 | `Sauron | Dunno, I can do echo test and time test |
| 01:53.01 | Delmar | well, what specifically is different about the Digitnetworks cards? its illegal for them to say they are 100% compat. when they are not... so what exactly is different? and I will get a reverse on my Visa. |
| 01:53.10 | `Sauron | but I never had anyone to call, or anyone to call me. |
| 01:53.11 | fearnor | it is 100% compatible. |
| 01:53.15 | fearnor | but it isn't x100p. |
| 01:53.17 | `Sauron | You know, internet people are scary ;) |
| 01:53.22 | Delmar | right. |
| 01:53.27 | fearnor | capiche? |
| 01:53.29 | netsurfer | greg_work fwd seems to be workin |
| 01:53.40 | greg_work | it used to constantly disconnect and reconnect for me |
| 01:53.50 | greg_work | haven't tried in a couple weeks though |
| 01:53.59 | *** join/#asterisk numBone (~numBone@c-24-129-204-233.se.client2.attbi.com) |
| 01:54.04 | `Sauron | weird |
| 01:54.14 | Delmar | anyway, I know people have issues with echo and line issues with these type of cards...original X100 or not. |
| 01:55.03 | netsurfer | a fix has been done but its not in the zaptel drivers |
| 01:55.10 | Delmar | there must be something wrong with my config somewhere. |
| 01:55.22 | Delmar | netsurfer oh what kinda fix? |
| 01:55.31 | netsurfer | Delmar - let me check.. found a page on it last nite |
| 01:55.33 | Delmar | netsurfer, for echo or other stuff? |
| 01:55.38 | Delmar | hey thank you... |
| 01:55.40 | netsurfer | for echo |
| 01:55.46 | Delmar | cool. |
| 01:56.26 | UncleBill | I'm having trouble finding a voip provider who I can hook an asterisk box to, and whom I can transfer a number to. Any suggestions or pointers? |
| 01:56.29 | Guest^DJ | non * questions: i existing have 4 PCs on a 4 ports router and is working fine. i plan to have another PC, how do i connect to the LAN without buying additional router ? i have a 24 port switch |
| 01:56.35 | Beirdo | ~seen mishehu |
| 01:56.38 | jbot | mishehu is currently on #asterisk (3d 13h 49m 56s). Has said a total of 8 messages. Is idling for 3d 7h 20m 41s |
| 01:56.45 | Beirdo | sigh |
| 01:56.48 | `Sauron | Unclebill: voicepulse seems to be able to do LNP |
| 01:57.03 | `Sauron | Guest: Can no do. |
| 01:57.15 | `Sauron | Err |
| 01:57.20 | `Sauron | you have a switch, put it inline |
| 01:57.44 | Dagrim | Hmmmm.. Yea I guyes SayNumber(1-xxx-xxx-xxxx) wouldnt work would it? lol |
| 01:57.48 | Dagrim | *guess |
| 01:57.55 | Dagrim | saydigits.. blah |
| 01:58.08 | *** join/#asterisk ionix (ionix@MTL-HSE-ppp201195.qc.sympatico.ca) |
| 01:58.11 | Guest^DJ | `Sauron: please adivce how |
| 01:58.24 | UncleBill | thanks, `Sauron. and they have no problem with me hooking up asterisk to them? others mandate you use their router and have a single instrument behind it. |
| 01:58.44 | Guest^DJ | Internet->router->switch->PCs ? |
| 01:58.56 | `Sauron | UncleBill: Look at their website. There's also VoicePulse Connect! which is perfect for * connections |
| 01:59.07 | UncleBill | thanks |
| 01:59.11 | `Sauron | I didn't like their pricing, and so I have to wait a month for my numbers to transfer over :( |
| 01:59.29 | Delmar | netsurfer, what I have tried so far is.. .changing a line in chan_zap.c ... #define READ_SIZE 160 to #define READ_SIZE 16 which I hear works better...but this didnt work one bit.... |
| 01:59.33 | fearnor | not only you are picky, you are also cheap |
| 01:59.35 | fearnor | heh |
| 01:59.44 | netsurfer | http://www.voip-info.org/tiki-index.php?page=Asterisk%20x100p%20echotraining u try this Delmar ? |
| 01:59.55 | Nugget | be aware that many people report poor service from voicepulse. their call quality seems to really vary depending on what area code your DID is in. |
| 02:00.18 | Nugget | I have two voicepulse DIDs and only one of them is usable. outbound calls are ok, though. |
| 02:01.55 | Delmar | netsurfer, yeah i have messed with all of that too. no gi. still massive echo. |
| 02:01.59 | Delmar | no go*. |
| 02:02.03 | *** join/#asterisk numBone (~numBone@c-24-129-204-233.se.client2.attbi.com) |
| 02:02.25 | Delmar | netsurfer, i googled for hours and didnt find much, and tried things i did find. |
| 02:02.45 | netsurfer | just found the page I was reading.. my mistake.. it was about UK CID |
| 02:03.00 | UncleBill | `Sauron, looks like voicepulse mandates you use a Sipura SPA-2000 that they sell you. are you sure they are Asterisk-able? |
| 02:03.07 | Delmar | netsurfer, i just tried that READ_SIZE thing suggested on another site and was sure that woulda been it but.. no gi. |
| 02:03.10 | netsurfer | Mark will not add the specific support for the X100P as "adds otherwise needless bloat to the zaptel side <-- heh |
| 02:03.12 | Nugget | UncleBill: http://connect.voicepulse.com/ |
| 02:03.45 | netsurfer | anyone here on sipgate.co.uk ? |
| 02:03.53 | netsurfer | my box wont register tonight :( |
| 02:04.18 | Delmar | I thought there was some kinda patch coming out to cure echo, which effects original cards as much as clones.... |
| 02:04.40 | netsurfer | are the clones worse? |
| 02:04.51 | netsurfer | my x100p clone is due on tuesday |
| 02:04.55 | Delmar | the same as I understand. |
| 02:05.01 | Delmar | where from? |
| 02:05.05 | netsurfer | uk supplier |
| 02:05.08 | Chuji | ~x100p |
| 02:05.10 | jbot | from memory, x100p is an obsolete card, copied by far too many people |
| 02:05.15 | *** join/#asterisk Rez (lorez@lorez.staff.freenode) |
| 02:05.30 | netsurfer | it may be obsolete, but its cheap and will do what I want it for |
| 02:05.35 | Delmar | lol its not obsolete. what are people supposed to byu instead? the 400 ? |
| 02:06.04 | netsurfer | why spend E180 when E20 will do the same thing |
| 02:06.16 | Delmar | indeed |
| 02:06.39 | Delmar | well i have an idea im going to go and play with....i know i can make this echo go away. |
| 02:06.51 | netsurfer | Delmar - if u do, let me know pls |
| 02:09.55 | UncleBill | Nugget and `Sauron: thanks. looks like they support what I want but they want $2.95/min for US long distance calls! ouch. |
| 02:10.02 | `Sauron | fearnor: |
| 02:10.04 | `Sauron | Cents |
| 02:10.09 | `Sauron | ignore fearnor |
| 02:10.13 | Nugget | uh, no, 0.0295. |
| 02:10.21 | `Sauron | Unclebill: VP connect is 2.95 cents/minute |
| 02:10.39 | UncleBill | uhhhh, yep. sorry about that |
| 02:10.48 | Nugget | just don't get too attached to the number they assign until you've confirmed that incoming calls are usable. |
| 02:11.14 | `Sauron | Ugh. I'm really not looking forward to porting 34 extensions to ast_data |
| 02:11.16 | `Sauron | fun fun |
| 02:11.27 | UncleBill | have you had issues with them, Nugget? |
| 02:11.44 | Nugget | yes, many people have. |
| 02:11.57 | UncleBill | that's no good then. |
| 02:12.01 | Nugget | some area codes are fine, some aren't. |
| 02:12.09 | Nugget | you might not have any problems |
| 02:12.09 | UncleBill | ok |
| 02:12.19 | Qwell | Basically the moral of the story is - test all providers before you use them |
| 02:12.24 | fearnor | basically |
| 02:12.28 | fearnor | you get what you pay for. |
| 02:12.35 | Nugget | I wish. |
| 02:12.43 | UncleBill | are there any other Asterisk-able carriers anyone can suggest? |
| 02:12.44 | Nugget | I'd be happy to pay for good service. :) |
| 02:13.07 | Nugget | I'm happy with nufone, but they don't offer local numbers, just tollfree numbers. |
| 02:13.14 | fearnor | one day pilosoft will do nationwide DIDs |
| 02:13.16 | fearnor | maybe. |
| 02:13.20 | ionix | sixtel |
| 02:13.21 | fearnor | :) |
| 02:17.53 | *** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com) |
| 02:24.16 | ManxPower | . |
| 02:24.38 | Guest^DJ | ManxPower: hi |
| 02:27.24 | beto75 | mikegrb? u here? |
| 02:28.02 | mikegrb | yes |
| 02:28.18 | beto75 | remember the bang bang bang |
| 02:28.23 | beto75 | was the DAMN Ulaw |
| 02:28.33 | beto75 | I change codec to GSM , and problem solved |
| 02:28.53 | greg_work | hm. nufone's webserver died, that instills a lot of confidence. interestingly, their page now contains almost as much information as before it crashed.... |
| 02:28.59 | beto75 | very strange |
| 02:29.32 | Dagrim | beto75: I cant even use gsm hardly.. speex is the only one that works decent on mine |
| 02:30.10 | beto75 | in your snom 200? |
| 02:30.15 | Delmar | mint. echo problem cured. |
| 02:30.24 | Dagrim | I dont use hardphones.. |
| 02:30.28 | beto75 | ahh |
| 02:30.32 | Dagrim | Im running a prepaid sys |
| 02:30.50 | Delmar | now all i gotta figure out is why the fark the cards are packing a sad during incomming calls. |
| 02:31.20 | beto75 | now excuse me guys, someone here has dealed with a welltech/micronet FXO sip device? |
| 02:31.50 | mikegrb | beto75: strange indeed |
| 02:31.56 | ManxPower | ~google site:lists.digium.com welltech fxo |
| 02:33.14 | beto75 | jbot: the very last says somethingabout my problem but no the solve |
| 02:33.28 | riksta | jbot, is a bot |
| 02:33.33 | Dagrim | lol |
| 02:33.38 | beto75 | AJJAJA |
| 02:33.50 | ManxPower | That's only the firest coupld of |
| 02:33.58 | beto75 | LOL |
| 02:34.00 | ManxPower | That's only the first couple of URLS matching the query. |
| 02:34.08 | riksta | it's not that funny |
| 02:34.09 | ManxPower | Go to google.com Do your own search |
| 02:39.35 | Delmar | man im having allot of trouble with these X100P's. i dont think anyone should buy one. they are a fuckin joke. |
| 02:40.00 | Delmar | im gonna buy a 400 and try that instead i think. |
| 02:40.00 | Dagrim | Maybe thats why theyre like $0.99 on ebay! |
| 02:40.03 | tzanger | Delmar: did you buy one from Digium, or a $10 from from ebay? |
| 02:40.11 | ManxPower | Delmar, Well if your telco impedenct is not 600 Ohms then it's kind of pointless anyway. |
| 02:40.25 | fearnor | impedance. |
| 02:40.26 | Dagrim | lol |
| 02:40.31 | Delmar | i know what u mean :P |
| 02:40.32 | CoaxD | made it to the home improvement stores to pick up 30 4' light fixtures today.. thank god they had 'em on sale because i sure needed 'em |
| 02:40.44 | Dagrim | Is there a difference between the ones on ebay and from digium?? |
| 02:40.47 | Delmar | no.. the echo thing is not a problem. that I have fixed really well. |
| 02:40.51 | tzanger | qty 30 four foot light fixtures? |
| 02:41.04 | CoaxD | Dagrim: Yeah. the ones on ebay don't support digium or asterisk in any way |
| 02:41.11 | tzanger | Dagrim: well digium's have support and are certified to work with asterisk and normal North American line impedances |
| 02:41.15 | Dagrim | ahhhh wow |
| 02:41.17 | Delmar | CoaxD put a sock in it.. lol |
| 02:41.24 | Dagrim | glad i didnt order like 10 of em like i was goin to =P |
| 02:41.43 | Guest^DJ | tzanger: damn, i was about to buy a few |
| 02:41.45 | CoaxD | Delmar: dude, dont ever even bother looking for support when they dont work |
| 02:41.58 | CoaxD | tzanger: Yeah, qty 30 four foot light fixtures |
| 02:42.05 | Guest^DJ | bought 3 from digitnetworks, and they work fine |
| 02:42.23 | CoaxD | GuestDJ: Dont let Mark hear you say that. he always blows a gasket when he hears that. |
| 02:42.35 | tzanger | CoaxD: what did you need so many light fixtures for? 4' light fixtures, you mean fluorescent bulbs? |
| 02:42.41 | Guest^DJ | CoaxD: :X |
| 02:42.42 | CoaxD | tzanger: Yeah, those are the ones |
| 02:42.47 | CoaxD | tzanger: I own an african violet nursrey |
| 02:42.49 | CoaxD | er nursery |
| 02:42.57 | tzanger | 120 linear feet of light, wtf are you doing, building a runway? |
| 02:43.02 | StilexIP | is anyone here using SER along with asterisk that would be able to help me really quick? |
| 02:43.04 | Dagrim | CoaxD: hear what? cussin? |
| 02:43.06 | tzanger | CoaxD: right on :-) |
| 02:43.08 | Delmar | 100% compatable... if the original X100p's work.. then these will work.... otherwise.. i can have the transaction reversed off my visa because Digitnetworks are advertising crap... |
| 02:43.19 | tzanger | Delmar: yeah you believe everything you read on ebay? |
| 02:43.29 | tzanger | I got Jimmy Hoffa's left shoe if you're interested |
| 02:43.36 | Delmar | fuck ebay, im talking about www.digitnetworks.com |
| 02:43.46 | Dagrim | hmm |
| 02:43.48 | Delmar | they are selling OEM X100P's. |
| 02:43.57 | tzanger | Delmar: so is Digium... :-p |
| 02:44.05 | ManxPower | The biggest difference between the Digium X101P/X100P and the generic clones is that you can still but the clones, you can't but the X100P/X101P from digium antmore (last I checked) |
| 02:44.14 | Delmar | and they state all over the place they are 100% compatable.. and will do anything the Digium cards will do. |
| 02:44.22 | Guest^DJ | is there a diff between digium X100P and diginetwork ?except for the price that is |
| 02:44.28 | Delmar | if thats not the case.. i will ring Visa and have the entire transaction reversed. |
| 02:44.31 | tzanger | Delmar: that's what they state... if they aren't working call 'em for support |
| 02:44.34 | ManxPower | Delmar, Um no. They are welling winmodems using the same chipset as the Digium card. |
| 02:44.44 | StilexIP | this is the problem - ser forwards the information to asterisk and I get this http://pastebin.ca/5007 |
| 02:45.01 | ManxPower | I bought two of them for $9 from some place like newegg and they work fine. |
| 02:45.23 | goatmilk | Delmar: i don't know what it is you are doing. but if you are building a foundation to something critical, you don't do it with crappy hardware. digium's stuff is good. |
| 02:45.29 | ManxPower | Of course since most of the systems I manage either have no FXO ports or more then 2 FXO ports we almost never use the X100P anyway |
| 02:45.32 | Delmar | ManxPower u have some from Digitnetworks working good? |
| 02:45.40 | tzanger | ManxPower: amen |
| 02:45.41 | Dagrim | Is today superbowl sunday? |
| 02:45.58 | ManxPower | Delmar, No, I bought the $9 windmodem from New Egg. Used the same chipset at the X100P |
| 02:46.09 | Dagrim | lol |
| 02:46.17 | Delmar | ManxPower yeah i heard u can do that :P |
| 02:46.27 | Delmar | but isnt the IO address written differently ? |
| 02:46.35 | ManxPower | All the X100P and the "clones" are are winmodems using a speicific chipset. |
| 02:46.35 | Delmar | so u have to modify the driver? |
| 02:46.55 | ManxPower | Delmar, kram put the PCI ids in the Zaptel stuff a LONG time ago for the clone cards. |
| 02:47.15 | Delmar | right.. so like I have been saying.. this whole... " oh.. you dont have an original Digium card? thats why... ".. those statements.. can fuck off. |
| 02:47.17 | ManxPower | I don't know why. I would have just told the people trying to use the clone cards "tough luck" |
| 02:47.39 | Delmar | yep. |
| 02:47.58 | Delmar | i noticed that when i grabbed the latest * CVS a while ago.. it had support for the clones. |
| 02:47.59 | ManxPower | But I ONLY use Digium equipment for everything except my home system |
| 02:48.19 | Delmar | im just havin some trouble with Asterisk and chan_zap not dealign with an incomming call properly... |
| 02:48.29 | Delmar | its getting confused and deciding there is a faulty line state ... |
| 02:48.34 | Delmar | it wasnt doing this before. |
| 02:48.37 | ManxPower | Now Digium went with a SPECIFIC winmodem card when they sold the X101P. |
| 02:48.38 | Delmar | so something is messed up. |
| 02:48.56 | ManxPower | Who know what might happen if you buy a card that uses the same chipset or a compatable chipset. |
| 02:49.41 | Delmar | so what is the chipset that Digium are using? |
| 02:49.47 | tzanger | http://it.slashdot.org/comments.pl?sid=137738&cid=11521466 |
| 02:49.48 | Delmar | or .. were. |
| 02:49.48 | tzanger | hahahahaha |
| 02:50.17 | *** join/#asterisk scrubb (~scrubb@OCI-19-41.OneCall.Net) |
| 02:51.03 | ManxPower | Delmar, I suppose I could look in my trash folder and find the message that was on the maliing lists about what chips each of the X10xP used, but that would take time. I think it was on asterisk-biz |
| 02:51.04 | Delmar | the Digitnetworks cards I have here, have a single fpga branded by Ambient. |
| 02:51.12 | Dagrim | tzanger: i dont get it =P |
| 02:51.26 | Delmar | bah. its not important. |
| 02:52.12 | ManxPower | Well THAT was easy! I just upgraded my SPA-841 from 2-line to 4-line |
| 02:58.59 | *** join/#asterisk Moc_ (~mochouina@modemcable212.49-80-70.mc.videotron.ca) |
| 03:00.34 | *** join/#asterisk Rez (lorez@lorez.staff.freenode) |
| 03:02.58 | okieplaya | some have sometime to help me get bellster up and goin i try every thing in the email still cant get it run? |
| 03:03.48 | K-Sensei | okie dokie, I traced the problem with the "Floating point exception" |
| 03:04.05 | K-Sensei | Anyone I can show the gdb output to who can tell me what to do about it? ;) |
| 03:04.26 | *** join/#asterisk NoRemorse (~me@202.161.68.6) |
| 03:04.34 | Qwell | K-Sensei: Could try posting it on pastebin |
| 03:05.10 | NoRemorse | hi all |
| 03:06.31 | K-Sensei | Okay here it is: http://pastebin.ca/5008 |
| 03:06.46 | K-Sensei | The same thing happens even if I just try to dial a music-on-hold extension |
| 03:07.18 | Chuji | ManxPower : What's your impressions of that phone? |
| 03:07.31 | Qwell | K-Sensei: You have to do a "bt" |
| 03:07.41 | K-Sensei | What is bt? |
| 03:07.47 | Qwell | backtrace, in gdb |
| 03:07.51 | K-Sensei | okay |
| 03:08.03 | K-Sensei | done, i will paste it to pastebin |
| 03:08.37 | K-Sensei | Updated: http://pastebin.ca/5009 |
| 03:08.43 | *** join/#asterisk VoicePulse (~VoicePuls@67.132.43.2) |
| 03:08.52 | freat | ManxPower: you did the software upgrade? |
| 03:08.57 | Qwell | Much better. :) |
| 03:09.32 | K-Sensei | It's something related to MOH for sure. |
| 03:09.32 | K-Sensei | I can run another trace with MOH only if that makes it simpler. |
| 03:10.47 | K-Sensei | obviously f->samples is 0. |
| 03:10.49 | *** join/#asterisk blankman (~blankman@h000d88a1570c.ne.client2.attbi.com) |
| 03:10.55 | blankman | Hey guys. |
| 03:10.56 | K-Sensei | so that causes a div by 0. |
| 03:11.30 | zigman | prolly |
| 03:12.25 | blankman | I have a question about the conditional includes ... as those parsed at dialtime or loadtime? Is one supposed to be able to conditionaly include a context based on parameters that are only known when a call has come in. |
| 03:12.25 | Qwell | ms could be some funky number too |
| 03:12.27 | Qwell | in theory |
| 03:12.41 | K-Sensei | does the / operator return a float or an int? |
| 03:13.01 | blankman | ~seen drkool |
| 03:13.02 | jbot | drkool <~drkool@210.211.144.70> was last seen on IRC in channel #asterisk, 8d 22h 41m 46s ago, saying: 'i am at my wit's end .Hoping some here can help'. |
| 03:13.18 | *** join/#asterisk Dagrim (~junglesto@dagrim.user) |
| 03:13.50 | justnulling | how can i setup simple config with two sip phones and call from one to another? |
| 03:14.04 | blankman | justnulling, look at the wiki |
| 03:15.08 | cypromis | ~seen snewpy |
| 03:15.12 | jbot | snewpy <~markl@203-206-235-209.dyn.iinet.net.au> was last seen on IRC in channel #asterisk, 13d 23h 33m 29s ago, saying: 'sskyles: and make the outgoing circuit in group one, and the incoming circuit in anything but group 1'. |
| 03:15.39 | evilbunny | cypromis: last i heard he's moving house |
| 03:15.47 | cypromis | ok |
| 03:17.00 | *** join/#asterisk mrproper_ (~mrproper_@61.95.55.242) |
| 03:17.54 | K-Sensei | okay I wrote a work-around & re-compiled |
| 03:17.55 | K-Sensei | testing... |
| 03:18.27 | Qwell | K-Sensei: You might want to file a bug report or something |
| 03:19.00 | K-Sensei | Could you point me to a guide on how to do it? |
| 03:19.13 | K-Sensei | Well, it's not crashing now, but it's not playing the hold music. |
| 03:19.13 | Qwell | bugs.digium.com |
| 03:19.25 | K-Sensei | I'm guessing for some reason it can't play the hold music - which is okay - but it shouldn't crash *. |
| 03:20.35 | Chuji | ~seen Chugee |
| 03:20.38 | jbot | i haven't seen 'chugee', Chuji |
| 03:20.59 | Chuji | ~seen Chuj1 |
| 03:21.00 | jbot | chuj1 <~b@68.52.145.41> was last seen on IRC in channel #asterisk, 184d 3h 42m 25s ago, saying: 'what on earth is backending bugs.digium.com? It's always so slow'. |
| 03:21.20 | Chuji | Heh, jbot has a good memory |
| 03:21.26 | Chuji | 184 days |
| 03:22.29 | K-Sensei | Cool, it works. |
| 03:23.28 | dan2 | kram: ping |
| 03:25.56 | K-Sensei | Qwell: Thanks for the direction. I have applied for an account on the bug tracker, and I'll post as soon as I get the password in e-mail. |
| 03:26.07 | Qwell | K-Sensei: It should be immediate |
| 03:26.19 | K-Sensei | I'm using hotmail, and it's not there yet. |
| 03:26.20 | Qwell | I had my email by the time I hit alt-tab to Thunderbird |
| 03:26.24 | Qwell | ahh :p |
| 03:27.29 | *** join/#asterisk okieplaya (~jjj@ip68-229-252-53.ok.ok.cox.net) |
| 03:29.41 | *** join/#asterisk BoRiS (~boris@S01060040ca1e5b54.wp.shawcable.net) |
| 03:36.09 | K-Sensei | Qwell: k, bug posted #3467 |
| 03:36.35 | blankman | Hey so does anyone else think that this should work: include => context_name|${ARG1}-${ARG2} |
| 03:36.39 | Qwell | K-Sensei: I don't really code, heh |
| 03:37.01 | *** join/#asterisk JohnJacob (~JohnJacob@pcp619824pcs.mainf01.in.comcast.net) |
| 03:38.15 | blankman | It doesn't, and from what I can see in the code, it is cause the includes are eval'd at extension loads ... meaning if you cli extension reload it looks to the extension.conf you have to have: include => context_name|8:00-17:30 ... ie concrete numbers .... anyone else think that it is important to have it evaluate on the fly? |
| 03:38.37 | blankman | This would be done in a macro off coarse :-) |
| 03:38.38 | Delmar | so whats the trick to get the X100 to detect that the calling party has hung up.. ie. they left a message on voicemail, and hung up? Currently, the x100 just holds the line open. |
| 03:39.18 | ManxPower | Delmar, "Kewlstart" signaling and a line that provides that signal. |
| 03:39.59 | Delmar | oh ok thats what the ks stands for. lol :P |
| 03:40.08 | Delmar | they are set fxs_ks |
| 03:40.18 | Delmar | and im sure out lines here in NZ supply all thats needed :) |
| 03:40.31 | Delmar | gotta be something wrong with my config somewhere... |
| 03:40.46 | ManxPower | Delmar, It needs US style far end disconnect supervision (indication) |
| 03:41.01 | Delmar | in english that is ? :) |
| 03:41.11 | netsurfer | FEDS ;) |
| 03:41.30 | Delmar | feds? where? |
| 03:41.32 | Delmar | :P |
| 03:41.34 | ManxPower | Unless NZ signals that the far end hung up in the same way the telcos in the USA do, kewlstart will not give yo anything. |
| 03:41.37 | netsurfer | :oP |
| 03:41.54 | Dagrim | eeek |
| 03:41.59 | Dagrim | who said feds |
| 03:42.02 | Dagrim | lol |
| 03:42.13 | goatmilk | what's wrong with us? |
| 03:42.35 | Delmar | caller hangs up...there is a click.. then.. .beep beep beep beep beep (by now u are a fuckwit for listening to the beeps for so long ) beep beep .. get the idea? :) |
| 03:42.59 | goatmilk | Delmar: you really should tone down the language in here |
| 03:43.24 | Delmar | sorry dude.. dont mean to curdle your milk :P |
| 03:43.44 | blitzrage | I love fast internet |
| 03:43.52 | Dagrim | mmmmhm |
| 03:44.16 | Delmar | so yeah.. i believe we have some basic .. signalling ManxPower, i have no idea if its the same as USA.. or whatever.... |
| 03:44.24 | K-Sensei | Qwell: I didn't think you were a coder, was just keeping you posted ;) |
| 03:44.30 | K-Sensei | Thanks again for your assistance. |
| 03:44.39 | Delmar | I know there are folk over here with this stuff all running mint... just don't know anyone that I can contact to get some help. |
| 03:44.43 | Qwell | K-Sensei: ahh, heh |
| 03:45.26 | ManxPower | Delmar, See Also: the horrible busydetect and busycount options in zapata.conf and read the mailing list archives for the issues with them |
| 03:46.51 | Delmar | yep. |
| 03:47.05 | Delmar | cheers for the input. ill get there i guess. :P |
| 03:47.38 | tangel | i have my house line plugged into asterisk |
| 03:47.42 | UncleBill | later, all |
| 03:47.51 | tangel | is there anyway to get both the ip phones and normal direct connect phones all ringing at the same time? |
| 03:48.52 | Nugget | yes. |
| 03:49.35 | blitzrage | Dial(SIP/1000&Zap/g2) |
| 04:00.48 | *** join/#asterisk harryvv (~noyb@S010600055d210201.vs.shawcable.net) |
| 04:03.19 | *** join/#asterisk jterrero (~some@ool-43576e0d.dyn.optonline.net) |
| 04:03.48 | okieplaya | if i cant get a public ip at home there some web site that i have seen that keeps track of your ip for you for a price |
| 04:03.55 | okieplaya | any one know the name of the site |
| 04:03.57 | okieplaya | ? |
| 04:04.21 | Delmar | nah for free |
| 04:04.24 | Delmar | dyndns.org |
| 04:05.07 | okieplaya | thanks you the man delmar will this work with a PBX? |
| 04:05.31 | Delmar | so you will get something like .. okieplaya.dyndns.org and create CNAME's for like.. pabx.yourdomain.com pointing to your okieplaya.dyndns.org domain. |
| 04:05.50 | harryvv | did a cvs of asterisk now command not found doing asterisk -c in root. |
| 04:05.54 | okieplaya | yes |
| 04:05.59 | harryvv | all directories are ther.e |
| 04:06.21 | Delmar | well.. kinda.. ok.. the box that u are running your pabx on .. will be on a private (ie 192.168 or 10.xx ) address? |
| 04:06.32 | Delmar | say.. behind a router .. that is doing NAT? |
| 04:06.38 | okieplaya | yes |
| 04:06.53 | okieplaya | it dont have to be i guess |
| 04:07.00 | okieplaya | it is now tho |
| 04:07.11 | jterrero | can someone help me out? i have a DID that i want to go to extension 100(main-menu), in extensions.conf i have something like this "exten => 7185552233,1,100() .... but debuging in the astgerisk cli tells me that 100 is not a valid application? |
| 04:07.40 | Nugget | that's because it is not. |
| 04:07.47 | Delmar | ok okieplaya, u need to forward some ports and it should work alright... i have issues and im going to make the Linux box have the public IP really soon tho.. so u might wanna consider that... |
| 04:08.04 | Nugget | in fact, I have no idea what you think you mean when you say "100(main-menu)" because that also makes no sense. |
| 04:08.10 | zigman | exten => 7185552233,s,goto(100) |
| 04:08.11 | Nugget | Goto() is what you want. |
| 04:08.17 | zigman | ir something |
| 04:08.18 | ManxPower | You ALWAYS need a priority |
| 04:08.40 | ManxPower | Goto(context,extension,priority) |
| 04:08.41 | Delmar | goto(context,extension) i think it is.. isnt it? |
| 04:08.47 | jterrero | thanks |
| 04:08.50 | Delmar | yup. i was close :P |
| 04:08.50 | okieplaya | Delmar what ports are thos |
| 04:08.53 | zigman | wasn't sure |
| 04:08.55 | zigman | it 5 am |
| 04:08.58 | zigman | sorry ;) |
| 04:09.00 | ManxPower | you could just "show appliction goto" as well ya know |
| 04:09.05 | Delmar | sec okieplaya ill get u the ports.... |
| 04:09.11 | okieplaya | thank you |
| 04:09.13 | zigman | or rtfm |
| 04:09.14 | zigman | ;) |
| 04:09.50 | Delmar | p[orts 5060, 4569, and 5004 UDP. |
| 04:10.25 | okieplaya | thank you will this help for SIP phone try in to get to my PBX? |
| 04:10.33 | Delmar | but i tell u what... i have major issues unless I do what some people call.. DMZ to the asterisk host. |
| 04:10.52 | okieplaya | yea thats what i have on now is DMZ |
| 04:11.08 | blankman | jterrero, If I am reading your question right, you dont want goto you want Dial. You aren't ask us how to goto priority 100 hundred in 7185552233, you are asking how you get 7185552233 to bridge to some other "phone" you have connected as extension 100 ... for that you use the dial command. You can find the info in the wiki. |
| 04:11.09 | Delmar | that is where say.. on an alcatel speed touch.. u set "defserver" .. which means.. any incomming traffic to any port whatsoever.. will forward to your DMZ host. |
| 04:11.11 | Delmar | thats the idea. |
| 04:12.09 | blankman | Anyone else besides me have the need to have an extension included or not included at dial time? |
| 04:13.20 | blankman | I can't seem to figure out how to code it ... I am trying to do it with compares, but he date/time values are string for the compare, so I can't do that ... and the include => context_name is an extension load time specific command :-( |
| 04:14.07 | blankman | I suppose I shouldn't call it extension and instead call it a context included at dial time. |
| 04:14.51 | *** join/#asterisk ta[i]nted (~ta_i_nted@65-60-70-242-cust.telepacific.net) |
| 04:16.16 | okieplaya | Delmar if I have DMZ turn to my PBX and someone trys to get to my PBX web server 192.168.1.101/maint they cant get there still no matter what port i FWD 80, and so on ? |
| 04:16.50 | okieplaya | and there useing public IP |
| 04:16.50 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 04:16.58 | okieplaya | not 192 |
| 04:17.54 | ariel_ | Hello everyone. |
| 04:17.58 | Nugget | hello ariel_! |
| 04:18.41 | okieplaya | hi |
| 04:18.42 | ariel_ | I have a question for someone. Does asterisk use /dev/dsp ??? It seems like it's in my RH9 and FC1 setups. |
| 04:19.10 | Nugget | it can, but nobody seriously uses that for anything. |
| 04:23.30 | *** join/#asterisk ccfiel (~chatzilla@210.213.138.81) |
| 04:25.26 | netsurfer | can anyone confirm if sipphone.com is working tonight? |
| 04:25.39 | netsurfer | I keep getting a message "please check your number" |
| 04:25.59 | `Sauron | No sipphone.com account |
| 04:31.05 | harryvv | whats a common reason when compiling asterisk for this error. looking in google right now. this error repeats near the bottom. its a cvs download. chan_zap.c :10030: error: dereferancing pointer to incomplete type |
| 04:33.13 | *** join/#asterisk static (abcbooze@adsl-218-242-245.jax.bellsouth.net) |
| 04:33.22 | static | hello! |
| 04:33.44 | static | i was wondering, what would be the cheapest way to set up 4 or 5 FXS lines? |
| 04:33.57 | static | or should i just go with sip phones? |
| 04:34.15 | beto75 | sipuras |
| 04:34.36 | static | i wish there was a cheap alterative |
| 04:34.37 | beto75 | with radio shack or wall mart analog phones |
| 04:34.43 | static | like with the x100p clones for fxo |
| 04:35.00 | beto75 | there are FXS cards from digium |
| 04:35.01 | static | sipuras are around $60/per unit right? |
| 04:35.18 | beto75 | well I think a little higher (AFAIK) |
| 04:35.30 | static | i think for the 1001 models |
| 04:35.34 | static | i might be wrong |
| 04:35.47 | static | i think thats what the grandstreama budgettone phones go for |
| 04:35.55 | static | granstream even |
| 04:35.58 | static | doh! |
| 04:36.10 | static | i give up on typing while smoking |
| 04:36.37 | netsurfer | give up on smoking its easier |
| 04:37.00 | static | heh |
| 04:37.02 | *** join/#asterisk JohnJacob (~JohnJacob@pcp619824pcs.mainf01.in.comcast.net) |
| 04:38.24 | beto75 | static: what does more expensive a hardphone (IP) is that you can not replace it quickly as a normal analog phone |
| 04:38.24 | riksta | is it possible, to match caller id's when they come through to make it display someones name, rather than their phone number on my cisco 79xx |
| 04:38.33 | riksta | like it does on a mobile |
| 04:38.36 | beto75 | sorry hardphone ( IP ) |
| 04:39.26 | static | hrmm |
| 04:39.28 | static | good point |
| 04:39.41 | static | i didnt consider if the phone takes a dive |
| 04:41.58 | netsurfer | "Please check your number - the user you are trying to reach is unknown" - anyone else heard this on sipphone.com ? |
| 04:42.46 | *** join/#asterisk datareactor (datareacto@203.81.192.33) |
| 04:44.41 | datareactor | can i use net2phone sip working with asterisk |
| 04:48.06 | *** join/#asterisk gopherspidey (~spidey@12-216-165-248.client.mchsi.com) |
| 04:51.14 | mrproper_ | anyone know why i would be getting "subscripted value is neither array nor pointer" compiling chan_capi? |
| 04:56.30 | riksta | anyone having problems logging onto sipgate? |
| 04:56.30 | *** join/#asterisk SexyKen (~sexyken@c-67-161-5-149.client.comcast.net) |
| 04:57.29 | netsurfer | yes |
| 04:57.36 | beto75 | guys I need to send a call to a fxo SIP gateway (unregistered to ASterisk) normally I send e164@host how do I put in extensions.conf Dial(sip/e164@host) ? |
| 04:57.56 | netsurfer | sipgate.co.uk is down riksta |
| 04:58.21 | netsurfer | :o\ |
| 04:58.26 | MrEntropy | in sip.conf, the name inside the square brackets [foo] is used as the username unless a username is specified underneath, is that correct? |
| 04:58.38 | netsurfer | yes |
| 05:00.05 | MrEntropy | is there any way to make an entity, where i can accept any username. like one context, for three phones, phone '111', '112', and '113' but they all use the same username and secret? |
| 05:00.11 | tangel | when i said: |
| 05:00.14 | tangel | is there anyway to get both the ip phones and normal direct connect phones all ringing at the same time? |
| 05:00.24 | netsurfer | careful using extn 112 |
| 05:00.29 | tangel | i meant the direct bell connected phones.. not stuff available on zap interfaces |
| 05:00.49 | MrEntropy | netsurfer: that was just an ex, i'm not really using it |
| 05:00.51 | tangel | like i dont want * to answer my bell line right now because i want the normal analog phones in my house to ring |
| 05:00.56 | beto75 | guys any help for me? |
| 05:00.59 | tangel | if that makes any sense |
| 05:01.04 | MrEntropy | netsurfer: is there a way i can do that? |
| 05:01.26 | netsurfer | MrEntropy - I dont think so.. why do u want to ? |
| 05:01.39 | *** join/#asterisk jskcr (~jskcr@jskcr.user) |
| 05:01.53 | riksta | is it possible, to match caller id's when they come through to make it display someones name, rather than their phone number on my cisco 7940 |
| 05:02.09 | netsurfer | MrEntropy - u can ring all 3 phones at once from a seperate extension f.ex. 114 |
| 05:02.13 | Nugget | riksta: yes. |
| 05:02.20 | riksta | Nugget: how please |
| 05:02.32 | MrEntropy | netsurfer: accept authed calls from multiple phones forwarded by ser sharing credentials |
| 05:02.34 | mrproper_ | anyone know why i would be getting "subscripted value is neither array nor pointer" compiling chan_capi? |
| 05:02.46 | Nugget | the cisco will automatically display the callerid name and number if available. |
| 05:03.02 | riksta | Nugget: ok, so i need to make a phonebook xml for it? |
| 05:03.11 | Nugget | however you can augment that with asterisk's built in callerid database which you can populate. |
| 05:03.15 | netsurfer | MrEntropy - sorry, outta my league there.. no idea |
| 05:03.19 | Nugget | no, it has nothing to do with the cisco xml menus. |
| 05:03.31 | riksta | Nugget: where is the built in caller db |
| 05:03.35 | `Sauron | Mmm. |
| 05:03.47 | Nugget | look into the asterisk command LookupCIDName |
| 05:03.51 | Nugget | it will explain |
| 05:04.02 | riksta | ok |
| 05:05.12 | netsurfer | one reason why not to depend on a free sip provider for your calls.. sipgate.co.uk down, I get to take tomorrow off work |
| 05:05.52 | SexyKen | Anyone know of a 2.0ghz Celeron with 512mb ram is enough horse to host a Nagios Server monitoring 40+ servers |
| 05:06.05 | *** part/#asterisk s[A]rumont (~sarumont@pcp04563795pcs.dallas01.ga.comcast.net) |
| 05:06.36 | Delmar | shoudl be plenty SexyKen |
| 05:06.51 | jskcr | More then enough. |
| 05:07.16 | `Sauron | SexyKen: More than enough. |
| 05:07.36 | SexyKen | So you think I should save my money and go with that then? |
| 05:08.01 | twisted | SexyKen, just go get a junker p133 |
| 05:08.04 | twisted | it should handle it just fine |
| 05:08.04 | riksta | nagios just runs a few little c and perl apps |
| 05:08.14 | riksta | to get snmp data |
| 05:08.55 | harryvv | anyone seen pbx_dundi break with lattest cvs? |
| 05:09.24 | Nugget | grande latte! |
| 05:11.01 | Guest^DJ | question: can i use net2phone hardware with * ? |
| 05:15.03 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 05:16.10 | Sedorox | Ok.. I'm at a loss... what files do I have to edit to get asterisk's boxes to link? just to link.. not even including dp swaps... |
| 05:17.04 | *** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc) |
| 05:19.54 | harryvv | netsurfer you here? |
| 05:20.07 | datareactor | can i use net2phone sip with asterisk |
| 05:21.06 | `Sauron | Sederox: iax.conf, I'd imagine |
| 05:21.22 | brc_ | datareactor, I doubt it |
| 05:21.26 | brc_ | ~nufone |
| 05:21.27 | jbot | nufone is probably Visit http://www.nufone.net for an excellent, native IAX termination service. |
| 05:22.47 | InfraRed | nufone is broken |
| 05:23.13 | DaLion | hey sauron |
| 05:23.33 | DaLion | http://www.nufone.net |
| 05:23.38 | `Sauron | DaLion buddy |
| 05:23.42 | DaLion | lol |
| 05:23.47 | DaLion | site down |
| 05:23.49 | shido6 | yep |
| 05:23.50 | `Sauron | nuphone.net didn't work for me |
| 05:23.52 | `Sauron | hehe |
| 05:23.53 | shido6 | drive took a shit |
| 05:23.59 | shido6 | we're fixing things |
| 05:24.04 | brc_ | `Sauron, eh? what's the problem? |
| 05:24.15 | `Sauron | brc: Nothing |
| 05:24.16 | brc_ | oh |
| 05:24.18 | brc_ | yeah I see |
| 05:24.38 | Sedorox | Jan 30 22:23:45 NOTICE[61490]: chan_iax2.c:5833 socket_read: Registration of 'stormy' rejected: Registration Refused |
| 05:24.56 | *** join/#asterisk makamani (~user@pub-nms.stcl.com) |
| 05:25.02 | Sedorox | Jan 30 22:24:35 NOTICE[89925]: chan_iax2.c:3871 register_verify: No registration for peer 'stormy' (from xxx.xxx.xxx.xxx) |
| 05:25.07 | Sedorox | what am I doing wrong? |
| 05:25.24 | Delmar | well there is your problem for a start... |
| 05:25.29 | Delmar | u are doing all this yesterday. :P |
| 05:25.46 | Delmar | its 6:30pm on the 31st right now :P |
| 05:26.03 | Sedorox | these are comps in Canada... MST... its 2 hrs behind.. so its the right time :-p |
| 05:26.08 | Sedorox | behind eastern... |
| 05:26.41 | Delmar | paste me the iax2 section you have for the client "stormy" |
| 05:26.48 | Sedorox | in what file? |
| 05:26.52 | Sedorox | iax.conf or sip.conf? |
| 05:27.02 | Sedorox | I really have no idea what I did... lol |
| 05:27.15 | Sedorox | just followed the examples and hoped it worked |
| 05:27.15 | Sedorox | :-p |
| 05:27.16 | Delmar | iax2.conf |
| 05:27.21 | Sedorox | oh.... maybe thats why |
| 05:27.30 | Delmar | well, whats is the client.. a SIP or IAX client? |
| 05:27.31 | Sedorox | I only have iax.conf |
| 05:27.51 | Sedorox | its two asterisk servers I'm trying to link.. so I figured I would do it iax |
| 05:28.00 | Delmar | yep. good idea. |
| 05:28.12 | Delmar | so u need to use iax.conf |
| 05:28.31 | Sedorox | ok.. thats where I have the stuff |
| 05:29.00 | Sedorox | [stormy] |
| 05:29.01 | Sedorox | type=peer |
| 05:29.01 | Sedorox | host=dymanic |
| 05:29.01 | Sedorox | context=* |
| 05:29.01 | Sedorox | secret=xxxxxxxxxxxx |
| 05:29.01 | Sedorox | trunk=yes |
| 05:29.09 | Sedorox | thats what is in iax.conf on the main box... |
| 05:29.27 | `Sauron | Sedorox: There's a page on voip-info.org on how to connect 2 * servers together |
| 05:29.35 | Sedorox | ok.. thanks ;-) |
| 05:29.38 | *** join/#asterisk Schism (~schism@clt74-101-230.carolina.rr.com) |
| 05:29.43 | *** join/#asterisk Othello (Othello@nusnet-156-21.dynip.nus.edu.sg) |
| 05:29.47 | `Sauron | http://www.voip-info.org/tiki-index.php?page=Asterisk+-+dual+servers |
| 05:30.11 | Sedorox | danka |
| 05:30.18 | jstorm | would someone else happen to use sixTel for outgoing? |
| 05:30.31 | *** join/#asterisk Inv_arp (junya@adsl-3-255-215.mia.bellsouth.net) |
| 05:30.35 | MrEntropy | can i make an entity entry in sip.conf that will authenticate on basis of ip, username, and password NOT the phone number/uri contact username |
| 05:30.36 | jstorm | international that is |
| 05:30.37 | MrEntropy | ? |
| 05:31.23 | evilbunny | MrEntropy: yes |
| 05:31.34 | evilbunny | [sipwhatever] |
| 05:31.39 | evilbunny | username=whatever |
| 05:31.44 | evilbunny | then in the dialplan |
| 05:31.44 | beto75 | guys how do I dial (extensions.conf) SIP e164@host |
| 05:31.56 | evilbunny | dial(sip/number@sipwhaterver) |
| 05:32.12 | mrproper_ | whats the best version of asterisk to compile with chan_capi-0.3.5 |
| 05:32.17 | evilbunny | beto75: www.e164.org has examples |
| 05:32.19 | MrEntropy | evilbunny: so 'sipwhatever' is a keyword? |
| 05:32.21 | beto75 | evilbunny and ther I can put also ${EXTEN} |
| 05:32.39 | evilbunny | MrEntropy: yup |
| 05:32.47 | MrEntropy | evilbunny: ok, cool |
| 05:32.51 | `Sauron | sipwhatever is the sip peer name |
| 05:32.57 | makamani | asterisk: hip.cpp:909: virtual void HipDataPCI::WriteDspSram(short unsigned int, short unsigned int, short unsigned int, word*): Assertion `0' failed. |
| 05:33.12 | makamani | am getting this error, anybody faced similar problems? |
| 05:33.41 | makamani | The line above says [chan_vpb.so] => (VoiceTronix V6PCI/V12PCI/V4PCI API Support) |
| 05:33.41 | makamani | <PROTECTED> |
| 05:33.45 | MrEntropy | evilbunny: where did you find out about this? it isn't in the wiki |
| 05:33.56 | evilbunny | MrEntropy: personal experience :) |
| 05:35.12 | MrEntropy | evilbunny: that's extensive |
| 05:35.51 | evilbunny | MrEntropy: little more then most, little less then others :) |
| 05:37.05 | *** join/#asterisk JohnJacob (~JohnJacob@pcp619824pcs.mainf01.in.comcast.net) |
| 05:39.39 | makamani | anybody facing this problem -> WARNING[11314]: loader.c:440 load_modules: Loading module chan_modem.so failed! |
| 05:40.05 | makamani | i fixed the above problem by rtfm. Needed to export VPB_MODEL=V12PCI |
| 05:41.31 | Sedorox | I'm still getting |
| 05:41.32 | Sedorox | <PROTECTED> |
| 05:42.02 | makamani | another error -> Ouch ... error while writing audio data: : Broken pipe |
| 05:42.10 | makamani | i m learning a lot it seems |
| 05:43.33 | jskcr | ~seen dagrim |
| 05:43.35 | jbot | dagrim <~junglesto@dagrim.user> was last seen on IRC in channel #asterisk, 1h 59m 43s ago, saying: 'mmmmhm'. |
| 05:44.04 | *** join/#asterisk wolfson (~hehe@bcp-68-187-181-159.man.nc.charter.com) |
| 05:44.08 | makamani | ok the above error of error whie writing audio data is due to asterisk dying. mpg123 is complaining that |
| 05:44.27 | makamani | but anybody with voicetronix cards getting the error of loading module chan_modem.so failed!??? |
| 05:45.15 | MrEntropy | evilbunny: http://pastebin.ca/5016 <-- this should do it, yes? match all calls no matter what the phone number, coming from a certain IP and with the same credentials |
| 05:45.59 | *** join/#asterisk makamani (~user@pub-nms.stcl.com) |
| 05:46.10 | makamani | but anybody with voicetronix cards getting the error of loading module chan_modem.so failed!??? |
| 05:46.27 | makamani | anybody facing this problem -> WARNING[11314]: loader.c:440 load_modules: Loading module chan_modem.so failed! |
| 05:46.32 | JamesDotCom | noload => chan_modem.so :D |
| 05:46.54 | makamani | ok |
| 05:47.23 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:48.07 | makamani | thanks JamesDotCom, have got passed that error now. Getting segmentation fault |
| 05:49.37 | JamesDotCom | haha |
| 05:49.40 | JamesDotCom | strace |
| 05:49.56 | makamani | thx again |
| 05:51.08 | Delmar | ok why would incomming calls have an echo problem .. and outgoing calls don't seem to have one... that is.. the SIP client has a bad self-echo receiving an incomming call via the pstn (fxo). |
| 05:51.18 | Delmar | outgoing calls seem great. |
| 05:51.26 | Delmar | so far anyway. |
| 05:52.48 | *** join/#asterisk peted20 (~pete@d4-81.rb.gh.centurytel.net) |
| 05:54.15 | makamani | it now keeps loading other moules. looks like i have to define, don't load default modules |
| 05:57.17 | *** join/#asterisk TheEmperor (~mattn@203.121.47.100) |
| 05:58.03 | MrEntropy | evilbunny: are you sure that's some sort of a keyword? |
| 05:59.23 | MrEntropy | evilbunny: i just searched the source code for it and i can't find a single instance. |
| 05:59.37 | paulfl | can anyone help with a feature group D wink problem? |
| 06:00.23 | paulfl | Does anyone know how to reverse or disable the wink for FEATD to work with a PBX not the telco. |
| 06:01.03 | evilbunny | lol |
| 06:01.09 | *** join/#asterisk djin (~marius@gridfox.xs4all.nl) |
| 06:02.36 | MrEntropy | evilbunny: =/ |
| 06:06.56 | evilbunny | MrEntropy: on the phone |
| 06:07.01 | evilbunny | gimme a sec |
| 06:08.18 | InfraRed | voip phone? |
| 06:08.19 | InfraRed | :) |
| 06:12.46 | netsurfer | Got 200 OK on REGISTER that isn't a register <-- what does this mean ? |
| 06:14.39 | MrEntropy | netsurfer: from what i've deduces you're probably getting packets from a previous registration after restarting asterisk |
| 06:14.51 | netsurfer | ok |
| 06:14.59 | netsurfer | that would explain this: That's odd... Got a response on a call we dont know about. |
| 06:15.09 | MrEntropy | yep =) |
| 06:15.33 | *** part/#asterisk ta[i]nted (~ta_i_nted@65-60-70-242-cust.telepacific.net) |
| 06:17.02 | Sedorox | <PROTECTED> |
| 06:17.06 | Sedorox | anyone know how to fix that? |
| 06:17.24 | netsurfer | set an allowed codec ? |
| 06:19.07 | Sedorox | I have allowed=gsm |
| 06:19.10 | Sedorox | er |
| 06:19.13 | Sedorox | allow=gsm |
| 06:19.15 | *** join/#asterisk clive- (~pirch@myw-stp-66-18-81-249.sentechsa.net) |
| 06:19.16 | Sedorox | but I still get that |
| 06:19.42 | jskcr | dissalow=all first |
| 06:19.55 | jskcr | err dam stuck s |
| 06:19.56 | makamani | i have extension local defined in extensions.conf |
| 06:20.07 | Sedorox | hmmm |
| 06:20.09 | makamani | it's a sample extensions.conf file |
| 06:20.11 | *** part/#asterisk beto75 (~beto75@201.128.177.84) |
| 06:20.20 | makamani | but still get -. Cannot find extension context 'local' |
| 06:20.23 | makamani | but still get -> Cannot find extension context 'local' |
| 06:20.35 | makamani | i said autoload=no in modules.conf |
| 06:21.09 | *** join/#asterisk shaZwaz (~chatzilla@203.81.196.167) |
| 06:21.24 | netsurfer | DEBUG[6108]: chan_iax2.c:3789 raw_hangup: Raw Hangup 213.208.106.212:53, src=4353, dst=257 <--- WTF!!! THATs my DNS server |
| 06:21.36 | shaZwaz | morning ppl |
| 06:21.59 | jskcr | :53 thats dns netsurfer |
| 06:22.12 | netsurfer | why is it flooding my asterisk ? |
| 06:22.37 | jskcr | Try running ethereal and setting a dst host and src host from the dns server |
| 06:23.04 | Sedorox | <PROTECTED> |
| 06:23.04 | Sedorox | Jan 30 23:22:51 WARNING[89925]: chan_iax2.c:7194 find_cache: Timeout waiting for ss-server:raucous3r5@stormy.smart-serv.net/local exten 1236 |
| 06:23.08 | Sedorox | .... *sighs* |
| 06:23.44 | Sedorox | why can't I connect to extentions across the connection? |
| 06:24.10 | Sedorox | I have disallow=all and allow=all in [general] in iax.conf |
| 06:24.53 | wasim | Sedorox: thats inane, why would you do that in iax.conf? |
| 06:25.12 | shaZwaz | morning wasim |
| 06:25.16 | Sedorox | for codecs? |
| 06:25.23 | wasim | Sedorox: generally, you'd disallow=all, and then allow=specificcodec |
| 06:25.24 | Sedorox | I dunno.. this is where I'm seeing it for iax connections |
| 06:25.26 | wasim | morning shaZwaz |
| 06:25.34 | Sedorox | thats that I have... |
| 06:25.36 | Sedorox | errr |
| 06:25.37 | wasim | Sedorox: yeah, like allow=gsm or something |
| 06:25.41 | Sedorox | sorry |
| 06:25.45 | Sedorox | I ment allow=gsm |
| 06:25.50 | Sedorox | *Sighs* its late.... |
| 06:25.53 | wasim | :) |
| 06:26.06 | Sedorox | but I don't understand why it still comes up with format unknown... |
| 06:26.15 | wasim | Sedorox: turn on iax2 debug, it seems as if the call is originating with some other format |
| 06:26.17 | Sedorox | and I get a busy signal with my phone telling me "404 " |
| 06:26.28 | Sedorox | hmmmmmm |
| 06:26.30 | Sedorox | true.... |
| 06:26.41 | Sedorox | but I even did allow=all.. and it didn't work |
| 06:27.19 | wasim | what about the far end ... whats the config there, and how is it getting the call |
| 06:27.47 | Sedorox | its just the samle setup... basiclly the same (I really haven't changed much on the one end, save extentions and added sip clients |
| 06:28.35 | ScythelX | hello all, Im trying to configure asterisk with ser but asterisk keeps outputting a ton of errors - could anyone look at this and maybe point me in the right direction.... http://pastebin.ca/5007 |
| 06:29.45 | Beirdo | fsp? who the hell uses fsp? |
| 06:31.01 | Sedorox | how do I turn on debugging of iax2? |
| 06:32.13 | visik7 | from the cli type iax2 debug |
| 06:32.59 | Sedorox | <PROTECTED> |
| 06:32.59 | Sedorox | <PROTECTED> |
| 06:32.59 | Sedorox | <PROTECTED> |
| 06:32.59 | Sedorox | <PROTECTED> |
| 06:32.59 | Sedorox | <PROTECTED> |
| 06:33.00 | Sedorox | <PROTECTED> |
| 06:33.03 | Sedorox | that format? |
| 06:33.48 | wasim | yeah ... that one |
| 06:34.09 | Sedorox | there is also later on in the log |
| 06:34.12 | Sedorox | er output |
| 06:34.13 | Sedorox | <PROTECTED> |
| 06:34.14 | Sedorox | <PROTECTED> |
| 06:34.18 | Sedorox | which is the same on both ends |
| 06:35.20 | Sedorox | hmmm |
| 06:35.45 | letherglov | odd...does it change later on in the log from the same source? |
| 06:36.23 | Sedorox | thats the last format that I see... on both console's.. and both are the same |
| 06:37.08 | Sedorox | know what.... |
| 06:40.49 | Sedorox | nope |
| 06:40.51 | Sedorox | still not working |
| 06:40.52 | Sedorox | same thing |
| 06:42.03 | freat | ahhhh.... finally got the QoS at work going awesome |
| 06:42.15 | *** join/#asterisk Dagrim (~junglesto@dagrim.user) |
| 06:42.18 | Dagrim | Heya |
| 06:42.29 | *** join/#asterisk harryvetch (~noyb@S010600055d210201.vs.shawcable.net) |
| 06:42.31 | freat | tested with a video stream and a huge download running at the same time, call quality was still good |
| 06:42.33 | Dagrim | Just got out of a meeting... good news.. |
| 06:43.03 | Dagrim | Im getting some 'traded' help as far as bandwith for my prepaid system.. |
| 06:43.22 | freat | ? |
| 06:43.39 | Dagrim | And Im getting help w/ advertising.. and people to sell pakcs of cards to.. |
| 06:43.48 | Dagrim | in return for 4 hours of work a week.. |
| 06:43.54 | Dagrim | on someone else's * system.. |
| 06:43.55 | Sedorox | cool |
| 06:43.56 | Dagrim | ;) |
| 06:44.13 | freat | you doing calling cards? |
| 06:44.15 | Dagrim | yea.. Im hyped.. |
| 06:44.16 | Dagrim | Yea |
| 06:44.25 | freat | sweet |
| 06:44.30 | Sedorox | *sighs* wish I could get this to work |
| 06:44.32 | Dagrim | Im actually gonna make $$ now |
| 06:44.47 | Dagrim | Because this town feeds off of college students.. thats all thats here.. |
| 06:44.55 | Dagrim | and 75% are foreign |
| 06:45.01 | Dagrim | And all buy calling cards.. |
| 06:45.12 | Dagrim | And I can offer them rates nobody else can.. and still make good $# |
| 06:45.51 | freat | you gonna get PRIs or pay for origination? |
| 06:46.09 | Dagrim | watcha mean.. |
| 06:46.15 | wasim | :) |
| 06:46.33 | Dagrim | Im still at stage 1 mind you =P |
| 06:46.40 | freat | how are people going to call into your * box? |
| 06:46.41 | djin | congrats, Dagrim. |
| 06:46.43 | Dagrim | The scripts I wrote work.. |
| 06:46.58 | Dagrim | Oh... a did .. 866 # |
| 06:47.03 | tangel | can i have * pick up the fxo line when i dial an extension? (i.e. answer the ringing line) |
| 06:47.15 | wasim | tangel: yes, Answer() |
| 06:47.33 | Dagrim | Im getting a freakin T1 for like '4 hours' of work. which is really only like 1 or 2 for me.. |
| 06:48.02 | Dagrim | And.. he knows places that will buy like 100 at a time.. |
| 06:48.02 | Dagrim | lol |
| 06:48.33 | Sedorox | <PROTECTED> |
| 06:48.34 | Dagrim | I used to work for him.. then things got sour.. he accused me of something that wasnt true.. anyways .., and he feels sorry for me I guess.. |
| 06:48.39 | Sedorox | <PROTECTED> |
| 06:48.50 | Dagrim | eh? unknown.. |
| 06:48.56 | Sedorox | <PROTECTED> |
| 06:49.02 | Sedorox | <PROTECTED> |
| 06:49.07 | Sedorox | I dunno whats going on... |
| 06:49.26 | Sedorox | and when I dial.. phone (Budgetone 100) comes up with 404... |
| 06:49.26 | tangel | wasim, how do i specify the port to answer? Answer(Zap/4) ? |
| 06:49.30 | wasim | Sedorox: thats not right ... something is screwy, in the specific iax.conf context, put disallow=all;allow=gsm |
| 06:49.32 | Sedorox | or if I already dialed it... 484 |
| 06:49.46 | Sedorox | kk |
| 06:49.58 | wasim | tangel: no, you specify that in s,1,Answer() in the [context] you define in zapata.conf |
| 06:50.23 | tangel | but i only want to answer it when i dial a specific extension |
| 06:51.06 | wasim | tangel: 77,1,Answer() |
| 06:51.25 | Sedorox | wasim: still getting the same errors with it in the defined blocks |
| 06:51.35 | tangel | just stick that under inbound-analog context? |
| 06:51.44 | tangel | and then dial 77 from any extension to pick it up? |
| 06:51.55 | wasim | Sedorox: something screwy, what * version are you using? |
| 06:52.16 | Sedorox | *CLI> show version |
| 06:52.16 | Sedorox | Asterisk 1.0.3 built by root@smart-serv.net on a i386 running FreeBSD |
| 06:52.16 | Sedorox | *CLI> |
| 06:52.23 | wasim | ugh ... |
| 06:52.26 | Sedorox | show version |
| 06:52.26 | Sedorox | Asterisk 1.0.3 built by root@stormy.smart-serv.net on a i386 running FreeBSD |
| 06:52.26 | Sedorox | *CLI> |
| 06:52.53 | *** join/#asterisk _m_ (~m@fbta199.fbta.uni-karlsruhe.de) |
| 06:53.04 | tangel | wasim, what am i missing here? :) |
| 06:53.39 | Sedorox | I (more or less) have this: [asterisk] |
| 06:53.39 | Sedorox | <PROTECTED> |
| 06:53.39 | Sedorox | <PROTECTED> |
| 06:53.39 | Sedorox | <PROTECTED> |
| 06:53.39 | Sedorox | <PROTECTED> |
| 06:53.40 | Sedorox | <PROTECTED> |
| 06:53.42 | Sedorox | <PROTECTED> |
| 06:53.44 | Sedorox | <PROTECTED> |
| 06:53.46 | Sedorox | <PROTECTED> |
| 06:53.48 | Sedorox | (cept name and stuff changed) |
| 06:53.49 | wasim | tangel: i really am not sure what youre trying to acheive, lets take it from the beginning |
| 06:54.07 | Qwell | Sedorox: You should really be using pastebin.com or something |
| 06:54.30 | Sedorox | sorry |
| 06:54.40 | tangel | my house line is plugged into * along with normal phones in the house |
| 06:54.40 | *** join/#asterisk terrapen (~cjs@cs662586-139.satx.rr.com) |
| 06:54.47 | terrapen | evening. |
| 06:54.56 | Dagrim | djin: Thanks on the congrats.. sorry I didnt see it up there ;-) |
| 06:55.05 | wasim | tangel: you have 1 fxo with the house line, and fxs with the phones? |
| 06:55.12 | tangel | i dont want * picking up the house line unless a * extension picks it up on purpose |
| 06:55.16 | terrapen | anybody remember the simple old Bell telephones of the 70s |
| 06:55.19 | terrapen | desk model. |
| 06:55.21 | tangel | that way the call goes to the normal answering machine,etc |
| 06:55.24 | terrapen | 12 buttons. |
| 06:55.36 | letherglov | terrapen, I have one |
| 06:55.40 | letherglov | the cortelco big old ones? |
| 06:55.41 | terrapen | handset cradled at the top, perpendicular to the phone |
| 06:55.42 | tangel | wasim, the phones i mentioned are normal house phones not plugged into * |
| 06:55.50 | letherglov | it's nice--great ringer |
| 06:56.00 | letherglov | used it as a wake up alarm connected to my fxs port for a while |
| 06:56.05 | letherglov | had the bios flip on the computer |
| 06:56.08 | terrapen | i'd really love to find a clone of these that had ethernet and IAX support |
| 06:56.09 | letherglov | cron initiate a test call |
| 06:56.14 | letherglov | worked well |
| 06:56.15 | tangel | i'm trying to get * and my pstn system to co-exist in harmony |
| 06:56.17 | terrapen | but otherwise looked and felt just like those old phones |
| 06:56.24 | terrapen | and no, i don't want to use an FXS |
| 06:56.30 | wasim | terrapen: we can make those for you |
| 06:56.44 | letherglov | yeah! just give wasim $30k |
| 06:56.47 | terrapen | i'd like an actually IP phone that looks and feels like the old 70s/80s bell models |
| 06:56.48 | wasim | tangel: but how can you tell * to pick up the link |
| 06:56.52 | Qwell | wasim: Who's "we"? |
| 06:56.55 | wasim | letherglov: no, no, we can do it much cheaper |
| 06:56.58 | letherglov | oh, heh |
| 06:57.01 | wasim | Qwell: farfon.com |
| 06:57.02 | letherglov | not the channel bank? |
| 06:57.04 | tangel | wasim, i guess i can't |
| 06:57.05 | letherglov | ;-) |
| 06:57.13 | letherglov | I suppose you've got the farfon pcb's now |
| 06:57.15 | tangel | i was hoping i could dial an extension and it would pick up the Zap/4 channel |
| 06:57.16 | wasim | tangel: bingo |
| 06:57.20 | letherglov | so you can just slap it in a new body |
| 06:57.26 | wasim | letherglov: yep, exacftly |
| 06:57.36 | brc_ | hola wasim |
| 06:57.46 | tangel | i think i could if Answer took an argument |
| 06:57.48 | wasim | or, you can just pick up a farfon, and strip it and plug it into a old deskset |
| 06:57.49 | letherglov | wel--what about |
| 06:57.57 | tangel | or if there was an OnHook rather than a Dial |
| 06:57.58 | letherglov | wasim's mod-your-craptacular-phone-extravaganza kit? |
| 06:57.59 | Sedorox | damn it.. still format unknown with allow=all.... wtf... |
| 06:58.07 | wasim | letherglov: exactly |
| 06:58.51 | wasim | evening mr brc_ |
| 07:00.23 | Sedorox | oi |
| 07:01.10 | terrapen | http://www.customphones.com/item80122.ctlg |
| 07:01.12 | tangel | is there any hardware that works in freebsd yet? |
| 07:01.13 | terrapen | that's the phone i want |
| 07:01.20 | terrapen | IAX ready, with ethernet |
| 07:01.24 | terrapen | wouldn't that be cool? |
| 07:01.29 | terrapen | those things are bomber. |
| 07:02.12 | wasim | terrapen: shouldn't be very difficult |
| 07:02.25 | terrapen | i guess i could hide an IAXy in one, eh? |
| 07:02.33 | wasim | terrapen: can you solder a little? |
| 07:02.42 | terrapen | yep, a little...im kind of shaky though |
| 07:03.10 | clive- | lol..wasim, what would one solder to an old phone?.:) |
| 07:03.16 | wasim | ok, then you just need to figure out the keypad matrix, and that should get you going |
| 07:03.22 | wasim | clive-: a farfon pcb, ofcourse |
| 07:03.31 | tangel | are there any decent 802.11 phones yet? |
| 07:03.57 | cypromis | no |
| 07:04.06 | terrapen | how will figuring out the keypad get me going? |
| 07:04.22 | clive- | WASIM, HOWS YOUR ATA'S COMMING ALONG |
| 07:04.27 | clive- | oops caps,. sorry |
| 07:04.38 | wasim | clive-: i have to send a relay to the dev bloke today, thats the only component left |
| 07:04.51 | tangel | cypromis, have you tried any? |
| 07:04.56 | cypromis | yes |
| 07:04.58 | cypromis | ad they all suck |
| 07:05.01 | tangel | i'd be willing to shell out 200$ if it was a nice phone |
| 07:05.01 | clive- | wasim, cool, |
| 07:05.36 | tangel | wlan600 is crap? |
| 07:05.36 | cypromis | try for yourself andhave fun |
| 07:05.36 | cypromis | :LD |
| 07:05.41 | tangel | i wish i knew a place i could try them :( |
| 07:05.43 | Sedorox | *Sighs* |
| 07:06.34 | tangel | a motorola engineer friend of mine is supposed to give me a phone he's been working on |
| 07:06.41 | tangel | gsm/802.11 w/ sip support |
| 07:06.57 | wasim | tangel: ask him if he can smuggle us a chipset or two with the datasheet, i'll make you one |
| 07:07.05 | terrapen | hmmmm...doesn't look like the IAXy supports PoE |
| 07:07.08 | Sedorox | lol |
| 07:07.09 | tangel | he tells me it only supports major network sip codecs though |
| 07:07.17 | terrapen | so i couldn't use one of them to build this phone |
| 07:07.20 | cypromis | inersting |
| 07:07.24 | cypromis | what is a major network sip codec ? |
| 07:07.26 | clive- | tangel sounds hot |
| 07:07.27 | terrapen | i only want one cable going to the phone |
| 07:07.29 | cypromis | is there an RFC for that ? |
| 07:07.44 | wasim | terrapen: now you're being picky |
| 07:08.06 | wasim | terrapen: use the two wire spare on the regular cable to power the iaxy |
| 07:08.07 | terrapen | the whole point of building this phone is high-tech retro |
| 07:08.21 | shaZwaz | wasim nice photo shoot on the farfon site |
| 07:08.22 | terrapen | and how retro can a phone be if it has to have a seperate power cable? |
| 07:08.26 | shaZwaz | :) |
| 07:08.34 | tangel | here's what it looks like: http://www.slashphone.com/70/642.html |
| 07:08.35 | cypromis | nah |
| 07:08.43 | cypromis | a phone with a dialpad is not relly retro |
| 07:08.53 | tangel | it runs the pocketpc kernel but not the UI |
| 07:08.56 | Qwell | wasim: Can you make me a rotary iax ip phone? |
| 07:08.56 | terrapen | well, maybe i will get one of the model 700's |
| 07:08.58 | terrapen | with the roatary |
| 07:09.01 | wasim | Qwell: affirmative! |
| 07:09.07 | Qwell | wasim: got any samples? heh |
| 07:09.12 | Qwell | I'd totally dig that |
| 07:09.19 | Sedorox | pretty looking |
| 07:09.20 | wasim | Qwell: we can even make one of those tall things, that just go onhook/offhook |
| 07:09.30 | terrapen | http://www.voip-info.org/wiki-Dial+Pulse+to+Touchtone+DTMF+Converters |
| 07:09.41 | terrapen | but that's just more bullshit |
| 07:09.43 | Qwell | wasim: I'd love to see an image of one |
| 07:09.44 | Sedorox | ooo ptt.. nextel.. here it comes... |
| 07:09.45 | Dagrim | wasim: Id buy something like that definetly.. look into it =) |
| 07:09.51 | terrapen | this all has to fit in the original phone's case |
| 07:09.58 | terrapen | or hidden somewhere |
| 07:10.06 | terrapen | Oh...maybe that's just it... |
| 07:10.08 | Sedorox | yea.. if it supports nextel.. which if the iden group did it.. it should.. let me know..... |
| 07:10.13 | terrapen | i just put the IAXy in another room |
| 07:10.21 | wasim | terrapen: :) |
| 07:10.25 | terrapen | or hidden under the cabinet |
| 07:10.35 | terrapen | no reason that the FXS has to be right next to the phone |
| 07:10.44 | terrapen | i'd love to have old retro phones |
| 07:10.52 | terrapen | and maybe even a retro Cobra cordless phone |
| 07:10.58 | niZon | put the fxs in the phone |
| 07:11.00 | niZon | or try |
| 07:11.07 | Qwell | wasim: what about an old Bell payphone? . . |
| 07:11.12 | terrapen | nizon, i'd love to but there is the power requirement and space constraints |
| 07:11.25 | Dagrim | terrapen: Thats a neat little device! |
| 07:11.35 | niZon | make an adaptor for power |
| 07:11.37 | *** join/#asterisk Chunder (~chatzilla@blk-222-123-93.eastlink.ca) |
| 07:11.38 | tangel | the hitachi wifi phone looks sweet |
| 07:11.50 | niZon | you have 2 spare pairs on the ethernet cable |
| 07:12.03 | Sedorox | *sighs* night... |
| 07:12.25 | Dagrim | hell yea it does |
| 07:12.31 | terrapen | i could do that |
| 07:13.03 | terrapen | now, how could i make my PBX sound/feel retro? |
| 07:13.21 | Qwell | terrapen: use words like "psychodelic" in your menus |
| 07:13.26 | niZon | lol |
| 07:13.26 | terrapen | what were the early PBXes like |
| 07:13.30 | Qwell | or "groovy" |
| 07:14.49 | terrapen | http://www.nwc.com/1220/1220ws1.html |
| 07:14.50 | wasim | terrapen: big, clicky |
| 07:14.57 | ScythelX | hello all, Im trying to configure asterisk with ser but asterisk keeps outputting a ton of errors - could anyone look at this and maybe point me in the right direction.... http://pastebin.ca/5007 |
| 07:18.09 | `Sauron | Dum di dum. |
| 07:18.15 | terrapen | http://www.cnn.com/2005/US/01/29/nazis.road.reut/index.html |
| 07:18.16 | terrapen | wahahah |
| 07:18.20 | terrapen | read the photo caption |
| 07:18.27 | terrapen | the Nazis have adopted a highway |
| 07:18.36 | terrapen | and put up the usual adoption sign |
| 07:18.40 | *** join/#asterisk labo (~ariel@route.flapping.net) |
| 07:18.49 | terrapen | and this guy tacks underneat it, a sign that says: |
| 07:19.00 | terrapen | "THEY ONLY PICK UP WHITE TRASH!" |
| 07:20.19 | terrapen | i don't think the IAXy would fit inside this phone |
| 07:20.30 | terrapen | the bell takes up too much space |
| 07:21.15 | clive- | has anyone used iaxcom? |
| 07:22.12 | Dagrim | Hey.. does anybody know if I get charged twice from nufone for: calling into the DID.. and then making a prepaid call out.. thats $.02 a min each way right? |
| 07:22.23 | terrapen | i guess i will put an IAXy in the phone closet and hook a wireless bridge up to it. |
| 07:23.06 | terrapen | i thot nufone allowed for free intra-provider calls...or was that voicepulse? |
| 07:23.20 | Chunder | anyone using any canadian iax termination? |
| 07:23.51 | terrapen | eh? |
| 07:24.00 | terrapen | (sorry, i couldn't resist...) |
| 07:24.38 | niZon | Chunder iax.cc has Canadian numbers |
| 07:24.38 | `Sauron | Humm. |
| 07:24.47 | Dagrim | How do I use voicemail() in a way that uses the personal greeting instead but only if the person recorded one? |
| 07:24.53 | `Sauron | ast_data isn't ready for primetime yet |
| 07:24.55 | `Sauron | Bummer. :( |
| 07:25.32 | Chunder | har! Really, I was just curious and looking around at pricing and such; I did see iax.cc thanks - no 902 numbers (Nova Scotia). pity. any others off the top of anyone's head? or maybe a toll-free number is the way to go if I can't get a local number? |
| 07:26.15 | niZon | I don't know of any that do 902 |
| 07:26.24 | niZon | I'm in Manitoba (204) |
| 07:26.31 | niZon | so I'm lucky |
| 07:27.18 | Chunder | I oculdn't find any either. wtf. we have phones here too. |
| 07:27.25 | *** join/#asterisk kks (~kks@203.115.208.140) |
| 07:28.03 | terrapen | are there many people in Nova Scotia? |
| 07:28.14 | Dagrim | chan_iax2.c:6600 socket_read: Received mini frame before first full voice frame |
| 07:28.18 | niZon | get a T1 and be the first to offer 902 DIDs :P |
| 07:28.20 | Dagrim | what is this? |
| 07:28.48 | niZon | Dagrim: did you try googling your errors? |
| 07:28.58 | terrapen | nizon, my thoughts exactly |
| 07:29.06 | terrapen | (t1) |
| 07:29.08 | Chunder | about a million |
| 07:29.10 | Dagrim | sorry! |
| 07:29.51 | niZon | Dagrim: np, check the wiki too |
| 07:29.59 | Dagrim | k |
| 07:31.37 | kks | if i using this exten => _8.,1,Dial(SIP/<ip>:5060,50,t) , do i need to register my *. but the destination isn't an asterisk box |
| 07:31.46 | *** join/#asterisk cdnpfunk (~chatzilla@pcp171275pcs.plsntv01.nj.comcast.net) |
| 07:32.18 | Chunder | yes, the t1 sounds like a great idea.... what's the going cost of a t1 line these days? |
| 07:32.45 | niZon | $800 or so? |
| 07:32.46 | Dagrim | I think you can get a remotely hosted box pretty cheap now. |
| 07:33.02 | Dagrim | If your using someone else.. but thats more $$$ |
| 07:33.45 | Dagrim | thats what Im thinking about doing .. maybe.. still toying with the idea |
| 07:37.36 | *** join/#asterisk JohnJacob (~JohnJacob@pcp619824pcs.mainf01.in.comcast.net) |
| 07:38.58 | Dagrim | I need a corporate logo so bad it isnt funny. |
| 07:39.08 | Dagrim | any ideas? =P |
| 07:39.23 | Dagrim | I was thinking something like.. |
| 07:39.26 | Dagrim | a phone.. lol |
| 07:40.26 | Dagrim | I have to admit.. * is insanely cool ;] |
| 07:41.38 | `Sauron | Hehe. |
| 07:42.04 | Dagrim | I used to work here.. what do you think of the design.. http://www.mychoice.net/ |
| 07:42.37 | netsurfer | having BIG problems making asterisk-addons, is this a mysql error? : http://pastebin.ca/5020 |
| 07:42.48 | Dagrim | They do VoIP phones to the middle of nowhere Southern Illinois .. heh |
| 07:42.51 | terrapen | <Dagrim> If your using someone else.. but thats more $$$ |
| 07:43.03 | terrapen | <PROTECTED> |
| 07:43.07 | terrapen | paste error |
| 07:43.13 | Dagrim | Lol.. what? |
| 07:43.23 | terrapen | does anybody else do that? |
| 07:43.28 | netsurfer | meeeeeeeeeeee |
| 07:43.30 | Dagrim | You messin with me? =P |
| 07:43.36 | terrapen | randomly select text in a window with the mouse cursor while reading? |
| 07:43.41 | netsurfer | yup |
| 07:43.46 | Dagrim | ya right |
| 07:43.48 | terrapen | i've been doing this forever |
| 07:43.57 | netsurfer | I do it to keep my clipboard fresh :D |
| 07:44.01 | terrapen | hahah |
| 07:44.05 | terrapen | anyway, i'm off to bed |
| 07:44.16 | netsurfer | g'nite |
| 07:44.38 | terrapen | night |
| 07:45.06 | *** join/#asterisk jerlique (~jerlique@lnk254.adl0.adsl.esc.net.au) |
| 07:45.50 | Dagrim | geeeeeze i didnt even realize i did that |
| 07:47.02 | `Sauron | terrapen: I do it to change the contrast. Dark background/light text makes for easier reading than light background/dark text |
| 07:47.07 | `Sauron | Hum |
| 07:47.09 | `Sauron | yeah, he left |
| 07:47.10 | Dagrim | lol |
| 07:47.20 | Dagrim | haha |
| 07:47.30 | `Sauron | Bummer that greg left |
| 07:47.41 | `Sauron | I recovered my * setup after playing with ast_data |
| 07:47.42 | Dagrim | well.. I copy a space to the clipboard sometimes to clear that buffer to PREVENT that from happening.. |
| 07:47.47 | Dagrim | Now I feel like a tard.. lol |
| 07:47.49 | `Sauron | hehe |
| 07:48.30 | Dagrim | what was wrong with your * ? |
| 07:48.39 | `Sauron | mine? nothing |
| 07:48.41 | *** join/#asterisk el_flynn (~el_flynn@219.95.108.54) |
| 07:48.47 | `Sauron | I was testing a new module for * |
| 07:48.54 | Dagrim | oh cool |
| 07:48.59 | `Sauron | I can safely say it's not ready for primetime. :( |
| 07:49.00 | Dagrim | whats the mod? |
| 07:49.03 | *** join/#asterisk gigel (~gigi@81.180.203.22) |
| 07:49.09 | inezk | `Sauron: what module? |
| 07:49.10 | netsurfer | `Sauron - u ever have probs doing make in asterisk-addons ? |
| 07:49.12 | gigel | hi all |
| 07:49.25 | `Sauron | dagrim: http://www.voip-info.org/tiki-index.php?page=Asterisk%20ast_data |
| 07:49.39 | `Sauron | netsurfer: Apparently I rarely run make in astersk-addons |
| 07:49.42 | `Sauron | asterisk-addons |
| 07:49.51 | netsurfer | uhm |
| 07:49.52 | `Sauron | None of those addons seem to interest me |
| 07:50.04 | netsurfer | ah thought thats where ast_data was |
| 07:50.06 | inezk | `Sauron: are you using some addons for mysql? |
| 07:50.07 | el_flynn | hello ast users |
| 07:50.14 | `Sauron | inezk: pgsql |
| 07:50.37 | inezk | `Sauron: which addons? |
| 07:50.46 | `Sauron | ast_data and chan_bluetooth |
| 07:51.16 | `Sauron | I need to see if Theo can give pointers to hack chan_bluetooth to use the headset profile instead of the hansfree profile |
| 07:51.29 | `Sauron | and ast_data needs some work with dialplan resolving stuff |
| 07:51.46 | Dagrim | `Sauron: Hmmmm.. so this makes things run a little smoother eh? |
| 07:51.59 | Dagrim | `Sauron: as far as db connections, etc etc |
| 07:52.07 | `Sauron | Dagrim: It lets you config everything from a DB - so you can make changes realtime on the fly, etc |
| 07:52.08 | *** join/#asterisk kks (~kks@203.115.208.140) |
| 07:52.20 | *** part/#asterisk djin (~marius@gridfox.xs4all.nl) |
| 07:52.26 | `Sauron | Seemed like the logical first step to making a decently-working web frontend to * |
| 07:52.27 | Dagrim | `Sauron: A db.. therefore a webmin easily. Right? |
| 07:52.33 | Dagrim | doh |
| 07:52.44 | Dagrim | Yeah.. Keep up the good work.. Thatd be killer. |
| 07:53.28 | `Sauron | hehe |
| 07:55.45 | *** part/#asterisk el_flynn (~el_flynn@219.95.108.54) |
| 07:56.46 | *** join/#asterisk r0d3nt|m (RatMan@64.60.114.35) |
| 08:01.34 | harryvetch | looks like im not getting red alerts when disconecting the phone line. |
| 08:04.45 | *** join/#asterisk Martohtar (Martohtar@194.19.32.100) |
| 08:06.07 | *** join/#asterisk tafazzi (~Dario@eporroj0-1.customer.nettuno.it) |
| 08:08.29 | jerlique | hi can I use a cisco 1600 as an interface into my BRI line from asterisk?? |
| 08:08.30 | *** join/#asterisk djin (~marius@62.58.40.196) |
| 08:13.29 | *** join/#asterisk cereal_ (~root@ngc2.uha.fr) |
| 08:13.38 | cereal_ | hello |
| 08:14.39 | cdnpfunk | i have a few questions regarding perl agi if anyone could pm and help me out |
| 08:15.40 | *** join/#asterisk r1 (~erwan@www.thiscow.com) |
| 08:16.12 | *** join/#asterisk cereal_ (~nico@ngc2.uha.fr) |
| 08:16.17 | cereal_ | hello |
| 08:16.36 | flewid | wussup |
| 08:16.49 | cereal_ | I m looking for someone good in ISDN config |
| 08:19.02 | *** join/#asterisk mitcheloc_ (~mitcheloc@69-169-28-46.anhmca.adelphia.net) |
| 08:23.07 | harryvetch | I have one channel configured for my zaptel yet get no redalarm when the phone line disconected. |
| 08:23.15 | *** join/#asterisk znoG (gs@200.115.216.109) |
| 08:23.53 | cereal_ | I dont manage to use chan_capi ; if someone could pm me |
| 08:25.08 | *** join/#asterisk mitcheloc_ (~mitcheloc@69-169-28-46.anhmca.adelphia.net) |
| 08:25.21 | *** part/#asterisk mitcheloc_ (~mitcheloc@69-169-28-46.anhmca.adelphia.net) |
| 08:25.56 | kks | hey, i'm trying to dial(SIP/x.x.x.x:5060,50,t), x.x.x.x is another sip compatible machine, but it appear that another side is busy. what is the problem, have anyone encountered this problem? |
| 08:26.08 | *** join/#asterisk Jas_Williams (~Jason@host217-43-52-65.range217-43.btcentralplus.com) |
| 08:27.23 | djin | cereal_, what's the problem? |
| 08:27.41 | cereal_ | djin can i pm you to explain the whole thing ? |
| 08:28.05 | djin | rather keep the discussing central. |
| 08:28.11 | cereal_ | ok |
| 08:28.13 | cereal_ | so : |
| 08:28.25 | cereal_ | got an AVM fritz card USB ; |
| 08:28.32 | cereal_ | configured correctly |
| 08:28.43 | cereal_ | chan_capi compiled and installed |
| 08:29.18 | cereal_ | srvasterisk*CLI> capi info |
| 08:29.18 | cereal_ | Contr1: 2 B channels total, 2 B channels free. |
| 08:29.31 | cereal_ | capi info can see it in the asterisk CLI |
| 08:29.35 | djin | looks good sofar |
| 08:29.37 | cereal_ | then in my dialplan |
| 08:30.29 | cereal_ | exten => 700,1,Dial(CAPI/B1:2) |
| 08:30.44 | cereal_ | 2 is an extensions dehind a ISDN PABX |
| 08:31.02 | djin | what's B1? |
| 08:31.18 | cereal_ | so when i do 700 on a phone for exemple the isdn phone 2 should ring and card should make channel D and B up |
| 08:31.27 | cereal_ | B1 is an ISDN channel isnt it ? |
| 08:31.44 | djin | did you configure capi.conf? |
| 08:31.54 | cereal_ | i dont know exactly the syntax for dialing with capi |
| 08:32.01 | cereal_ | yep here is capi . conf |
| 08:32.12 | djin | wait. |
| 08:32.19 | djin | plese use pastebin.ca |
| 08:32.29 | cereal_ | ? |
| 08:32.38 | djin | http://pastebin.ca |
| 08:32.45 | djin | don't flood the IRC |
| 08:32.59 | cereal_ | ok |
| 08:33.47 | *** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com) |
| 08:34.01 | *** join/#asterisk pranav (dawda_pran@203.115.89.135) |
| 08:34.31 | pranav | hello |
| 08:34.44 | pranav | hi to everyone on this channel |
| 08:35.02 | cereal_ | djin : http://pastebin.ca/5023 |
| 08:35.14 | pranav | i am facing some problem in configuring asterisk |
| 08:36.21 | pranav | i have configured the sip.conf as well as the sip phones but the dialtone is not coming |
| 08:36.22 | djin | cereal, better change context in something like 'capi' |
| 08:36.29 | cereal_ | ok |
| 08:36.32 | cereal_ | doing it |
| 08:36.46 | pranav | hi djin |
| 08:37.06 | djin | then you can define a [capi] in extensions.conf to handle incoming calls. |
| 08:37.41 | *** join/#asterisk JohnJacob (~JohnJacob@pcp619824pcs.mainf01.in.comcast.net) |
| 08:37.48 | djin | based on this try Dial(CAPI/@8300,2,60,r) |
| 08:38.00 | djin | hi pranav |
| 08:38.07 | pranav | ya hi |
| 08:38.11 | cereal_ | djin my problem is to go outside |
| 08:38.20 | cereal_ | to call another number for example |
| 08:38.34 | djin | this means you dial '2' on capi-device '8300' |
| 08:38.48 | djin | I understand, try the Dial command I just posted. |
| 08:39.06 | pranav | ok so you are going out, can we talk for 5 min |
| 08:39.22 | cereal_ | ok let me try that |
| 08:39.40 | djin | oh, made a mistake |
| 08:39.56 | djin | it's Dial(CAPI/@8300:2,60,r) |
| 08:39.56 | jerlique | Does anyone know if I use a cisco 1600 as an interface into my BRI line from asterisk?? (Like a channel bank) |
| 08:42.05 | cereal_ | djin tried |
| 08:42.22 | cereal_ | when i make 700, it s like busy |
| 08:42.32 | cereal_ | on the isdn phone behind pabx |
| 08:44.45 | pranav | i have configured the phone as a sip client, do i need to register that in sip.conf |
| 08:45.31 | djin | cereal, you try to use an isdn phone as client on the Asterisk PABX? |
| 08:46.11 | cereal_ | djin : there is a pabx (alcatel) linked to the USB fritz card |
| 08:46.21 | cereal_ | behind this pabx are phones |
| 08:46.28 | djin | ah, ok |
| 08:46.31 | pif | the mwi doesn't flash on my polycom 600 (I followed voip-info's instructions), known problem? |
| 08:46.34 | djin | similar setup here |
| 08:46.35 | cereal_ | for exemple i tested that : |
| 08:46.43 | *** join/#asterisk adrianhensler (~chatzilla@blk-222-123-93.eastlink.ca) |
| 08:47.01 | cereal_ | with no asterisk i made a capifax 1200 file.txt and that worked when i put a fax at 1200 extensions |
| 08:47.14 | pranav | hello adrianhensler |
| 08:47.18 | cereal_ | so the link and the alcatel is ok |
| 08:47.22 | adrianhensler | hello |
| 08:47.23 | djin | this looks like a configuration between the Alcatel and Asterisk in accepting calls. |
| 08:47.29 | pranav | can we talk |
| 08:47.33 | djin | This Dial command should work. |
| 08:47.49 | pranav | i have a small problem in configuring the sip phone |
| 08:47.55 | cereal_ | i cpoy pasted your Dial command .. just put 1200 instead of 2 |
| 08:47.58 | djin | you could also try replacing the 2 with an outside number to test. |
| 08:48.12 | adrianhensler | I won't be any help.... |
| 08:48.24 | pranav | why |
| 08:48.40 | djin | cereal, just to make sure, you pasted the second one right? With the : instead op ,? |
| 08:49.08 | adrianhensler | because I have never set up a sip device |
| 08:49.20 | pranav | i just want to ask whether we have to register the sip phone if we are using it as asip client |
| 08:49.27 | cereal_ | exten => 700,1,Dial(CAPI/@8300:1200,60,r) |
| 08:49.31 | *** join/#asterisk mitcheloc_ (~mitchel@69-169-28-46.anhmca.adelphia.net) |
| 08:49.35 | cereal_ | this one is ok ? |
| 08:49.41 | djin | yes, look ok. |
| 08:49.53 | mitcheloc_ | could someone tell me if this dropped packet looks like it could be sip related? cause i can't get any incoming BV calls to work! |
| 08:49.54 | mitcheloc_ | Jan 31 00:42:53 main kernel: IN=eth0 OUT= MAC=00:04:5a:29:2d:ec:00:50:57:00:8e:ff:08:00 SRC=202.182.15.147 DST=69.169.28.46 LEN=908 TOS=0x00 PREC=0x00 TTL=116 ID=1460 PROTO=UDP SPT=8488 DPT=1028 LEN=888 |
| 08:50.10 | djin | what happens if you replace the 1200 with an external number to test connectivity? |
| 08:50.13 | *** join/#asterisk Trionnis (buffy@12-203-113-15.client.insightBB.com) |
| 08:50.38 | cereal_ | the Alcatel simulates an outide sumber |
| 08:50.45 | cereal_ | 1200 is like an external phone |
| 08:50.52 | Trionnis | would anyone here be familiar with /contrib/asterisk-ices.xml ? |
| 08:50.59 | *** part/#asterisk brc_ (~brian@brc.base.supporter.pdpc) |
| 08:51.10 | Trionnis | or just icecast conference streaming in general? |
| 08:51.46 | pranav | hi Trionnis |
| 08:51.51 | Trionnis | 'lo |
| 08:51.55 | Trionnis | how goes it? |
| 08:51.57 | Trionnis | :) |
| 08:52.08 | djin | cereal, I think you have to do some testing with the link with Alcatel. F.e. use a normal S0 phone to test dialing 1200 and see if that works. |
| 08:52.21 | pranav | i am sorry i dont know,infact i wanted to ask you something |
| 08:52.30 | djin | Your Asterisk config looks ok now |
| 08:53.01 | cereal_ | djin : just did it |
| 08:53.08 | pranav | hello djin can we talk now |
| 08:53.08 | cereal_ | phone can call |
| 08:53.20 | djin | hmmm. |
| 08:53.46 | cereal_ | no i try that with the phone behind alcatel : i call the 8300 ( the usb modem on asterisk) the D channel IDSN comes up but no B channel |
| 08:54.05 | cereal_ | is there a way the 1200 --> 8300 make a ip phone behind asterisk ring ? |
| 08:54.15 | djin | sure. |
| 08:54.24 | cereal_ | is it a deflection ? |
| 08:54.38 | cereal_ | i m not really familiar with ISDN s... |
| 08:54.38 | djin | Asterisk needs to know it has to pickup. |
| 08:54.55 | djin | and handle the call according context capi in extensions |
| 08:55.11 | cereal_ | ok so |
| 08:55.29 | djin | My capi.conf has isdnmode=multipoint, but I not sure if that changes things |
| 08:55.37 | cereal_ | in extensions.conf ; [capi] i put a Dial(SIP/umber) |
| 08:55.43 | djin | yes |
| 08:56.02 | cereal_ | ok |
| 08:56.15 | cereal_ | but how does asterisk know when a call arrives he must forward it ? |
| 08:56.41 | djin | If it meets requirements like extension, etc. |
| 08:56.47 | djin | Similar to Zap. |
| 08:57.00 | *** join/#asterisk int19h (Miranda@219.95.162.48) |
| 08:57.20 | int19h | howdy all... |
| 08:57.33 | pranav | hi djin |
| 08:58.01 | cereal_ | djin this is in my extensions.conf : |
| 08:58.02 | pranav | hi int19h |
| 08:58.09 | cereal_ | [capitest] |
| 08:58.09 | cereal_ | exten => s,1,Dial(SIP/ns,20) |
| 08:58.16 | int19h | am writing an asterisk app module... was wondering: is it possible to access variables in the dialplan from apps? |
| 08:58.22 | int19h | without passing them as params I mean... |
| 08:58.29 | mitcheloc_ | hey does anyone know what this might mean? Call-ID: 6d6f9d23298044214ce568442578418c@127.0.0.1 |
| 08:58.43 | mitcheloc_ | does that mean it's registering but saying my ip is localhost? |
| 08:58.44 | djin | cereal, is capitest the context you have in your capi.conf? |
| 08:58.56 | cereal_ | yep |
| 08:58.59 | djin | ok |
| 08:59.07 | *** join/#asterisk wolfson (~hehe@bcp-68-187-181-159.man.nc.charter.com) |
| 08:59.23 | djin | and calls to 1200 within alcatel go to you USB? |
| 09:00.04 | djin | well, try a reload and see what happens. |
| 09:01.00 | *** join/#asterisk Delvar (~irc@83.146.53.34) |
| 09:01.15 | harryvetch | who here uses x100p |
| 09:03.04 | cereal_ | djin OK i can call from 1200--> 8300 |
| 09:03.11 | cereal_ | just the other way still is fucked .. |
| 09:03.13 | flewid | i use x100p |
| 09:03.18 | Trionnis | ok, * doesn't seem to be running ices |
| 09:03.21 | Trionnis | the module is there |
| 09:03.28 | Trionnis | the stream extension is being created |
| 09:03.43 | djin | pranav, stop addressing people directly. Just post your question. |
| 09:04.16 | djin | cereal, as least that's something. |
| 09:04.23 | Trionnis | I just don't see it running, and icecast doesn't show a source stream |
| 09:04.34 | cereal_ | maybe it s this Alcatel that sux |
| 09:04.40 | Trionnis | anyone help please? :) |
| 09:04.43 | pranav | ya i configured the sip.conf as well as the sip phones but i am not getting the dialtone |
| 09:04.46 | cereal_ | university hardware ... |
| 09:05.13 | djin | cereal, tied * to a Lucent here. |
| 09:05.21 | djin | same config. |
| 09:05.41 | datareactor | ~datareactor |
| 09:05.57 | djin | cereal? |
| 09:06.11 | pranav | actually i have pasted my sip.conf in the pastebin |
| 09:06.30 | djin | did you say that 1200 was the extension on the Alcatel to the * ISDN? |
| 09:06.35 | pranav | pastebin.ca/5025 |
| 09:07.32 | *** join/#asterisk denon (denon@synapse.subneural.net) |
| 09:07.32 | *** mode/#asterisk [+o denon] by ChanServ |
| 09:08.02 | cereal_ | bbl |
| 09:08.16 | djin | cereal? |
| 09:08.23 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
| 09:09.19 | slePP | A-Tuin: are you a giant turtle? |
| 09:09.22 | slePP | with elephants'n'all? |
| 09:10.22 | pranav | mr.djin please help me |
| 09:10.28 | cereal_ | yep djin ? |
| 09:10.41 | djin | did you say that 1200 was the extension on the Alcatel to the * ISDN? |
| 09:10.49 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 09:10.55 | cereal_ | 1200 is the ISDN phone behind the Alcatel |
| 09:11.15 | djin | and 8300 is the extention of asterisk? |
| 09:11.33 | cereal_ | 8300 is the ISDN port linked to asterisk via the USB card true |
| 09:11.41 | djin | ah, ok |
| 09:12.02 | cereal_ | i tested 5 min ago : 1200-->8300 works ( redircted on a sip phone behind the asterisk ) |
| 09:12.06 | djin | thought you were calling yourself for a second ;) |
| 09:12.49 | djin | yes, pranav? |
| 09:13.21 | *** join/#asterisk bjohnson (~bjohnson@ip226-181.tor.istop.com) |
| 09:13.35 | pranav | ya djin i have configured the sip phone as well as the sip.conf accordingly |
| 09:13.48 | Delvar | and good morning to you all |
| 09:14.35 | pranav | but the dialtone is not coming |
| 09:14.54 | djin | calling from the sip phone, you mean? |
| 09:15.39 | pranav | means i have only as yet 1 sipura phone which i configured to my asterisk |
| 09:16.19 | djin | you mean you can call TO the sipura, but not FROM the sipura? |
| 09:16.19 | pranav | i have pasted my sip.conf n the pastebin.ca/5025 |
| 09:17.16 | djin | you're missing a [from-sip] context to handle the calls from you sipura. |
| 09:17.19 | pranav | no actually till now i have only i sipura phone which i have configured but, i am not getting the dialtone in that |
| 09:17.53 | pranav | ok so i need bracket to sip-conf |
| 09:18.23 | pranav | sorry from-sip |
| 09:18.43 | *** join/#asterisk humblast (~serty@212.247.174.226) |
| 09:19.15 | djin | you need to setup a from-sip contect in extensions. |
| 09:20.05 | pranav | so what change i have to do in extensions.conf |
| 09:20.31 | djin | pranav, this is basic configuration. Read the docs. |
| 09:21.42 | pranav | ok but do i need the registration line in my sipconf |
| 09:22.53 | djin | registration line? |
| 09:23.43 | pranav | some where i read register => 1234 user@mysipprovider.com/1000 |
| 09:23.53 | pranav | this was in some site |
| 09:24.20 | djin | no. |
| 09:25.26 | pranav | ok so i hvae made context=[from-sip] in extensions.conf |
| 09:25.57 | *** join/#asterisk meppl (~mephisto@pD9E69488.dip.t-dialin.net) |
| 09:26.50 | *** join/#asterisk Syncros (~sysop@noc.routermonkey.net) |
| 09:27.16 | pranav | do i need to make any more changes |
| 09:35.27 | pranav | yes mr. djin |
| 09:38.12 | *** join/#asterisk JohnJacob (~JohnJacob@pcp619824pcs.mainf01.in.comcast.net) |
| 09:39.54 | *** join/#asterisk Makenshi (~makenshi@2001:630:1c0:2001:280:c8ff:fee2:921f) |
| 09:48.02 | cereal_ | djin style here |
| 09:48.16 | djin | yes |
| 09:48.24 | cereal_ | checked Alcatel PBX |
| 09:48.34 | *** part/#asterisk MrEntropy (~entropy@ppp136-158.lns1.adl2.internode.on.net) |
| 09:48.41 | *** join/#asterisk MrEntropy (~entropy@ppp136-158.lns1.adl2.internode.on.net) |
| 09:48.43 | cereal_ | still dont manag to place a call from what is behind asterisk to what is behind Alcatel |
| 09:49.05 | MrEntropy | where in the sip message should username and password be? if they're to be sent i plaintext? |
| 09:49.07 | djin | but do you manage to place calls to outside Alcatel? |
| 09:50.52 | cereal_ | manage to call from phones behind alcatel ==> phones behind asterisk |
| 09:50.56 | cereal_ | but not the other way |
| 10:00.25 | *** join/#asterisk denon (denon@synapse.subneural.net) |
| 10:00.25 | *** mode/#asterisk [+o denon] by ChanServ |
| 10:02.12 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
| 10:03.07 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 10:03.21 | puzzled | morning |
| 10:09.02 | *** join/#asterisk h3x (~Justino@nv-65-40-157-57.sta.sprint-hsd.net) |
| 10:13.45 | *** join/#asterisk ellvis (~ellvis@adsl-data-237.84-47-64.telecom.sk) |
| 10:13.49 | ellvis | hi people |
| 10:18.39 | *** join/#asterisk kaitseb (~sadie@193.17.41.120) |
| 10:20.22 | humblast | A little question about the codecs used in asterisk: There are some header files in codecs folder who's names ends with _ex.h ... when I take a look at them I am dumbfounded, I have no clue as to what they are about, does anybody know? |
| 10:23.09 | zyke | has anyone irish DIDs ? |
| 10:23.24 | tafazzi | I have a biproc xeon 3Ghz handling no more than 150 concurrent calls on pure SIP. Does this sound ok or can I get more tuning kernel or asterisk? |
| 10:24.27 | jerlique | tafazzi - I dont know the answer- but out of curiosity what happens when you try to dial the 151'st call? |
| 10:24.56 | tafazzi | You star missing calls for timeout problem. |
| 10:25.04 | tafazzi | Call drop... |
| 10:25.10 | tafazzi | random . |
| 10:25.21 | tafazzi | CPUs at 80% |
| 10:26.00 | *** join/#asterisk tsimshatsui (~BBRodrigu@pD9EA62EB.dip.t-dialin.net) |
| 10:26.19 | tsimshatsui | hi everyone, after inserting "progressinband=yes" into sip.conf, * still does not send 183 to originating gateway, there is no ringback tone, anyone help please ? |
| 10:28.05 | *** join/#asterisk Davetaz (zatevad@sown-86.ecs.soton.ac.uk) |
| 10:28.32 | Davetaz | hey all :) |
| 10:28.47 | Davetaz | anyone know if there is any interest in supporting IPv6 in asterisk? |
| 10:31.07 | zyke | tafazzi: are you using transcoding when you get to 150 calls? |
| 10:31.13 | *** join/#asterisk adnans (~adnans@noterik2.demon.nl) |
| 10:32.07 | *** join/#asterisk ckruetze (~ckruetze@i528C21FB.versanet.de) |
| 10:33.04 | tsimshatsui | hi everyone, after inserting "progressinband=yes" into sip.conf, * still does not send 183 to originating gateway, there is no ringback tone, anyone help please ? |
| 10:34.29 | tafazzi | zyke, we have two scenarios... The hardest is using transcoding. g729 -> gsm. |
| 10:37.13 | clive- | tafazzi, you need a dsp card imho |
| 10:37.28 | *** join/#asterisk sambal (~sambal@gateway.office.flatbox.nl) |
| 10:38.04 | *** join/#asterisk JohnJacob (~JohnJacob@pcp619824pcs.mainf01.in.comcast.net) |
| 10:42.02 | zyke | tafazzi: how many calls can you handle when you do transcoding? |
| 10:51.55 | tafazzi | zyke, I'm doing transcoding... |
| 10:52.02 | tafazzi | g729->gsm |
| 10:52.18 | *** join/#asterisk thefallen (PolarBear@thefallen.user) |
| 10:53.26 | Mavvie | wonder what an overlap call is. |
| 10:54.31 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
| 10:56.35 | evilbunny | overlap or overflow? |
| 10:56.41 | Mavvie | overlap |
| 10:58.11 | Jas_Williams | Overlap dialling is where digits are passed on at atime rather than in one message |
| 10:58.23 | Jas_Williams | one at a time |
| 10:58.58 | Mavvie | oh, that's from level 5 all the time. |
| 10:59.01 | Mavvie | that makes sense. |
| 10:59.10 | Mavvie | thanks jas. |
| 10:59.15 | Jas_Williams | No problem |
| 11:02.27 | *** join/#asterisk los415 (~los415@adsl-67-127-57-254.dsl.pltn13.pacbell.net) |
| 11:06.18 | tafazzi | zyke, I'm at 150 concurrent calls. |
| 11:08.16 | MrEntropy | how is the username and password put in the sip message for auth? |
| 11:09.15 | pashah | hello everybody |
| 11:09.31 | pashah | im trying to use zapbardge and getting 'app_zapbarge.c:173 conf_run: Error setting conference' |
| 11:09.35 | pashah | anybody? |
| 11:10.33 | zyke | tafazzi: that's a good number.. |
| 11:11.02 | zyke | i'm lookin to buy a box to support around 80 concurrent calls |
| 11:11.17 | zyke | sip to sip and sip to iax2 calls |
| 11:11.47 | tafazzi | Thank you zyke... In reality the number is 300. Because the machine is doing g729->sip from FireFly clients to another Asterisk IVR. |
| 11:12.31 | tafazzi | 80 concurrent calls... |
| 11:12.40 | tafazzi | From PSTN to SIP? |
| 11:13.25 | zyke | if you only do IVR you should be able to do quite a lot as opposed to having long sip to sip conversation |
| 11:23.33 | *** join/#asterisk jpmcallister (~jpmcallis@kapla.escelsa.com.br) |
| 11:24.27 | *** join/#asterisk bowman (~bowman@snert3.tal.de) |
| 11:25.12 | bowman | is there a way to redirect manager events to a file? |
| 11:27.07 | *** join/#asterisk Negrizprovod (~iwn@yurik.nis.nnov.su) |
| 11:27.24 | jpmcallister | Has anyone ever seen this message: |
| 11:27.26 | jpmcallister | Auto fallthrough, channel 'IAX2/2001@2001/2' status is 'UNKNOWN' |
| 11:27.52 | jpmcallister | It is happenig after a Background message |
| 11:28.52 | jpmcallister | If a press a digit while the message is playing, everything works and I am redirected to the correct exten. If I hear the entire message, the line hungs up and I get this message on console |
| 11:29.18 | jpmcallister | s/If a/If I/ |
| 11:31.46 | *** part/#asterisk Negrizprovod (~iwn@yurik.nis.nnov.su) |
| 11:36.51 | *** join/#asterisk mAsH` (~mAsH@host46-29.pool8173.interbusiness.it) |
| 11:36.54 | mAsH` | hi all |
| 11:37.42 | *** join/#asterisk r0d3nt|m (~RatMan@4.19.77.194) |
| 11:38.20 | mAsH` | anyone can help me...i cannot compile asterisk :/ |
| 11:38.41 | *** join/#asterisk maruz (~maumar@adsl-123-3.38-151.net24.it) |
| 11:39.09 | *** join/#asterisk JohnJacob (~JohnJacob@pcp619824pcs.mainf01.in.comcast.net) |
| 11:39.13 | *** join/#asterisk tih (~tih@athene.hamartun.priv.no) |
| 11:39.35 | zyke | guys.. any one got an irish DID? |
| 11:39.37 | pashah | mAsH` which errors u get? |
| 11:41.12 | mAsH` | my gcc... |
| 11:41.26 | mAsH` | configure: error: installation or configuration problem: C compiler cannot create executables. |
| 11:41.26 | mAsH` | make: *** [editline/libedit.a] Error 1 |
| 11:41.41 | mAsH` | i nevere had an erroro like this... |
| 11:41.48 | pashah | OS |
| 11:41.49 | pashah | ? |
| 11:42.03 | mAsH` | slack 10 |
| 11:42.10 | mAsH` | kernel 2.4.26 |
| 11:42.40 | pashah | gcc -v |
| 11:42.42 | pashah | ? |
| 11:42.51 | mAsH` | if i try to install the gcc 3.4 i get the same error |
| 11:43.06 | mAsH` | gcc version 3.3.4 |
| 11:43.33 | pashah | configured with:... |
| 11:43.53 | mAsH` | pardon |
| 11:43.53 | mAsH` | Reading specs from /usr/lib/gcc-lib/i486-slackware-linux/3.3.4/specs |
| 11:43.53 | mAsH` | Configured with: ../gcc-3.3.4/configure --prefix=/usr --enable-shared --enable-threads=posix --enable-__cxa_atexit --disable-checking --with-gnu-ld --verbose --target=i486-slackware-linux --host=i486-slackware-linux |
| 11:43.53 | mAsH` | Thread model: posix |
| 11:43.53 | mAsH` | gcc version 3.3.4 |
| 11:45.30 | pashah | imho u should have something like --enable-languages=c,c++,java,f77,pascal,objc,ada,treelang |
| 11:45.43 | mAsH` | :/ |
| 11:46.43 | zyke | guys.. i'm looking for incoming irish numbers... can anyone help? |
| 11:47.00 | mAsH` | i have just finished to install slack |
| 11:47.25 | mAsH` | * is the first app that i'm trying to complie |
| 11:47.50 | *** join/#asterisk ckruetze_ (~ckruetze@i3ED61127.versanet.de) |
| 11:47.51 | pashah | mAsH`: |
| 11:47.55 | pashah | try export CC=gcc |
| 11:48.17 | mAsH` | in Makfile? |
| 11:48.40 | pashah | nope just enter from keyboard: export CC=gcc [press enter] |
| 11:48.48 | pashah | and then make |
| 11:48.56 | mAsH` | opsss... |
| 11:48.57 | mAsH` | ok |
| 11:49.48 | mAsH` | idem |
| 11:49.50 | mAsH` | :/ |
| 11:56.10 | cereal_ | i m desesperating |
| 11:56.21 | cereal_ | fucking isdn card |
| 11:58.39 | *** join/#asterisk zotz (~zotz@24.244.133.136) |
| 12:00.04 | *** join/#asterisk libpcp (libpcp@210.16.20.5) |
| 12:02.09 | libpcp | hi all |
| 12:02.31 | libpcp | is it possible to setup 2 outbound provider in a dialplan? |
| 12:04.16 | zotz | libpcp, why not? (I am not an expert around here) I see no problems |
| 12:04.44 | *** join/#asterisk ckruetze (~ckruetze@i3ED61127.versanet.de) |
| 12:05.24 | cereal_ | anyone here s quite good with chan_capi ? |
| 12:05.49 | *** join/#asterisk McKillroy (~mckillroy@L0957P13.dipool.highway.telekom.at) |
| 12:05.55 | McKillroy | Hello ! |
| 12:06.00 | libpcp | zotz: can you give me a sample for that? |
| 12:06.07 | Jas_Williams | cereal_: do you need a hand |
| 12:06.25 | zotz | libpcp, what are you trying to do? |
| 12:06.38 | McKillroy | Does anyone know if there is a possibility to use the Cell phone Bluetooth profiles to transmit voice to * ? |
| 12:07.15 | libpcp | zotz: i want that when the provider 1 is not available, the call will use the provider 2 |
| 12:07.21 | ellvis | hi |
| 12:07.47 | zotz | ah, not sure if i can give such an example easily. let me see |
| 12:07.49 | *** join/#asterisk mac_7 (~karsten@c149212.adsl.hansenet.de) |
| 12:07.55 | ellvis | i am looking for some txt based sip phone, does exist anything like that?:) |
| 12:08.02 | cereal_ | Jas_Williams yep :) |
| 12:08.40 | ellvis | i am not sure if linphonec work with asterisk |
| 12:08.44 | zotz | libpcp, is provider one voip and 2 pots or 1 and 2 both voip? |
| 12:10.19 | cereal_ | ok dont movve |
| 12:11.28 | libpcp | both voip |
| 12:12.50 | libpcp | zotz: my purpose of that because my 2 accounts on both voip provider call only use one at a time. so if theres a call on voip provider 1, the call will switch to voip provider 2 on the dialplan |
| 12:14.11 | *** join/#asterisk X-DBA (~cfmcginni@user-0c99m7l.cable.mindspring.com) |
| 12:14.19 | *** part/#asterisk X-DBA (~cfmcginni@user-0c99m7l.cable.mindspring.com) |
| 12:14.24 | pashah | libpcp: transfer on busy to next provider |
| 12:15.20 | *** join/#asterisk adnans (~adnans@noterik2.demon.nl) |
| 12:15.22 | pashah | libpcp: using increase by 100 thing |
| 12:16.11 | libpcp | pashah: something like that.. |
| 12:16.14 | libpcp | exten => _1.,1,Dial(SIP/voip1/${EXTEN}&SIP/voip2/${EXTEN}) |
| 12:16.19 | libpcp | is this possible? |
| 12:16.48 | pashah | libpcp: should work, the first one wins |
| 12:16.57 | pashah | the first that answers |
| 12:17.06 | Delvar | pashah: yes that should work |
| 12:17.07 | evilbunny | erm not a valid url is it? |
| 12:17.20 | evilbunny | sip/${EXTEN}@voip1 |
| 12:17.21 | evilbunny | etc |
| 12:17.26 | libpcp | and if the first 1 is not available it will go to voip2 ? |
| 12:17.41 | evilbunny | both will ring at the same time |
| 12:17.43 | Delvar | evilbunny: SIP/entity/${EXTEN} works too |
| 12:17.53 | evilbunny | entity? |
| 12:18.05 | Delvar | whatever teh entity is in sip.conf [this bit] |
| 12:18.13 | libpcp | both will ring at the same time? |
| 12:18.20 | Delvar | libpcp: yes |
| 12:18.40 | pashah | libpcp: after one is answered the ringing will stop |
| 12:18.47 | libpcp | so what will happened if both provider answered the call ? |
| 12:18.58 | Delvar | first come |
| 12:19.03 | evilbunny | unlikely to be a race condition |
| 12:19.21 | libpcp | so is it safe to use that dialplan for a multiple provider? |
| 12:19.27 | pashah | libpcp: yep |
| 12:19.30 | Delvar | yes, i do it all teh time |
| 12:20.11 | *** join/#asterisk pulu (~chatzilla@65.77.78.3) |
| 12:20.38 | libpcp | thats nice :) |
| 12:20.49 | libpcp | will thanks a lot guys.. |
| 12:20.55 | evilbunny | libpcp: would be interesting if someone has voicemail and your first call causes the 2nd to go to voicemail and then because it answered you end up with only voicemail |
| 12:21.53 | libpcp | evilbunny: how will i do that in the dialplan ? |
| 12:22.27 | evilbunny | the way i do it is something like... |
| 12:22.32 | zotz | libpcp, back - still checking |
| 12:22.45 | evilbunny | [macro-provider] |
| 12:22.46 | evilbunny | exten => s,1,Dial(${ARG1}@astman.aus-biz.com,60) |
| 12:22.46 | evilbunny | exten => s,2,Hangup |
| 12:22.46 | evilbunny | exten => s,102,Macro(pstn,${ARG1}) |
| 12:22.46 | evilbunny | exten => s-NOANSWER,1,Hangup |
| 12:22.48 | evilbunny | exten => s-BUSY,1,Congestion |
| 12:22.50 | evilbunny | exten => s-CHANUNAVAIL,1,Macro(pstn,${ARG1}) |
| 12:22.52 | evilbunny | exten => s-CONGESTION,1,Macro(pstn,${ARG1 |
| 12:23.00 | evilbunny | so if the first fails, but it wasn't a connect it'll try again |
| 12:25.15 | *** join/#asterisk mcisse (~mcisse@ARennes-303-1-5-111.w80-15.abo.wanadoo.fr) |
| 12:27.17 | libpcp | evilbunny: that config is only for single provider? |
| 12:27.27 | libpcp | and it something like a redial? |
| 12:28.24 | *** join/#asterisk ellvis (~ellvis@adsl-data-237.84-47-64.telecom.sk) |
| 12:28.24 | ellvis | re |
| 12:38.10 | *** join/#asterisk Aze` (~Aze_@host229-162.pool80105.interbusiness.it) |
| 12:38.10 | Aze` | hi all |
| 12:39.23 | *** join/#asterisk JohnJacob (~JohnJacob@pcp619824pcs.mainf01.in.comcast.net) |
| 12:39.42 | Aze` | Anyone can help me with Sip ? |
| 12:41.11 | tih | ~docs |
| 12:42.05 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 12:43.14 | *** join/#asterisk wasim_ (~wasim@203.81.200.8) |
| 12:44.33 | Aze` | my problem is: i cant listen when use Modem/ttyI0, why ? |
| 12:44.34 | *** join/#asterisk mrverizone (~mrverizon@pa-robinson1b-88.pit.adelphia.net) |
| 12:49.30 | *** join/#asterisk lasseman (Lars@195.66.41.17) |
| 12:51.12 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 12:54.03 | lasseman | Does anyone know of a method to setup voicemail so that users are auto created...kind of like autocreatepeer in sip.conf? Access to the asterisk is limited by firewalls and so on, so only authorized users will be able to connect. |
| 12:57.40 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 12:59.40 | MrEntropy | does asterisk accept plaintext auth? |
| 13:01.30 | *** join/#asterisk Negrizprovod (~Negrizpro@yurik.nis.nnov.su) |
| 13:03.29 | *** join/#asterisk multrix (~chatzilla@ALyon-110-1-12-235.w81-48.abo.wanadoo.fr) |
| 13:04.17 | multrix | hi everybody, I'm looking for a good Pabx frontend (web is better) and simple, witch do you think I should take ? there are a lot ! |
| 13:04.29 | multrix | s/pabx/asterisk/ |
| 13:04.59 | *** part/#asterisk Negrizprovod (~Negrizpro@yurik.nis.nnov.su) |
| 13:07.04 | *** join/#asterisk thieumS (~darkmind@nanterre-7-82-229-210-142.fbx.proxad.net) |
| 13:07.54 | thieumS | hello there |
| 13:08.17 | thieumS | does anybody use ast_data addon ? |
| 13:11.35 | *** join/#asterisk Othello (Othello@nusnet-156-21.dynip.nus.edu.sg) |
| 13:13.29 | *** join/#asterisk Manipura (~chatzilla@dsl-ep-209-115-250-i114-cgy.nucleus.com) |
| 13:13.50 | *** join/#asterisk pick_a_nick (mikeeeee@mikes.TerraNet.Ro) |
| 13:15.36 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 13:17.24 | Emrah | thieumS: Bonjour |
| 13:18.34 | Emrah | multrix: Salut :) |
| 13:18.57 | multrix | ah un francais ! |
| 13:19.06 | Emrah | Un suisse!!! |
| 13:19.11 | Emrah | ;) |
| 13:19.17 | multrix | ah pardon ;) |
| 13:19.27 | multrix | t expert en asterisk ? |
| 13:19.27 | Emrah | Joke :) |
| 13:19.28 | multrix | :) |
| 13:19.51 | Emrah | Je suis administrateur de plusieurs solutions Asterisk pour des providers |
| 13:20.08 | Emrah | Que recherches-tu? |
| 13:20.34 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 13:21.25 | multrix | en fait j'suis apprenti ingénieur et mon projet c trouver une solution technique et commerciale pour la voip dans ma boite |
| 13:21.36 | multrix | pour l'instant on fé de l'install reseau, c une ptite boite francaise |
| 13:21.50 | multrix | mais on voudrai ajouter des cordes a notre arc |
| 13:21.57 | multrix | et la voix sur ip s'est présentée |
| 13:22.05 | multrix | et moi j'aimerai argumenter pour proposer de l'asterisk |
| 13:23.06 | Emrah | L'aventage de l'Asterisk est que les coûts sont vraiment réduits, tant au niveau de l'installation que des appels sur l'extérieur. |
| 13:23.17 | Emrah | De plus, c'est un logiciel fabuleux en constante évolution. |
| 13:23.40 | multrix | c ce que j'y vois... mais dans une boite ou pour l'instant j'lui le seul a peu pres competant pour me lancer dans l'apprentissage d'asterisk |
| 13:23.46 | multrix | tu penses k'ya un modele commerical possible ? |
| 13:23.55 | multrix | vendre des ptites boites qu'on mette dans les pme |
| 13:24.01 | multrix | qui interface avec l'autocom classique |
| 13:24.10 | multrix | et permette une migration progressive vers la VoIP ? |
| 13:24.15 | Emrah | Quel type de ligne vous possédez? |
| 13:24.22 | multrix | chez nous ? |
| 13:24.28 | Emrah | oui |
| 13:24.30 | multrix | nous on a trois PRI |
| 13:24.35 | Emrah | ISDN, E1, ... |
| 13:24.36 | multrix | T2 ca s'apelle en france |
| 13:24.51 | multrix | c 2B+D |
| 13:24.58 | multrix | si je me souviens bien |
| 13:25.01 | multrix | on en a 3 |
| 13:25.06 | Emrah | Alors c'est une ligne ISDN standart |
| 13:25.34 | multrix | atten jdis des conneries je crois |
| 13:26.11 | Emrah | Ce qui serait envisageable de réaliser ce serait: installer un serveur asterisk, y connecter une carte qui emploierait le boîtier isdn... En réseau interne, il suffira de connecter des périphériques hardware ou software qui utiliseront Asterisk. |
| 13:26.57 | *** join/#asterisk vaewyn (freeman@mail.deltamach.com) |
| 13:27.29 | multrix | on a trois T0 |
| 13:27.35 | multrix | mais moi je parlais pas pour nous en interne |
| 13:27.44 | multrix | masi plutot un service a proposer a des clients |
| 13:27.55 | vaewyn | baaaah |
| 13:29.14 | multrix | :D |
| 13:29.19 | *** join/#asterisk jetscreamer (~jetscream@adsl-64-219-216-41.dsl.hstntx.swbell.net) |
| 13:30.27 | *** join/#asterisk r1 (~erwan@www.thiscow.com) |
| 13:31.38 | thieumS | oula ça cause technique la :p |
| 13:32.04 | junky[work] | multrix: tu peux parler anglais? |
| 13:32.27 | Emrah | multrix: IL serait intéressant de passer par un provider externe dans ce cas |
| 13:32.39 | Emrah | 3 t0 sont largement insuffisants |
| 13:33.07 | thieumS | je suis provider IP --> Telephonie publique |
| 13:33.09 | *** join/#asterisk r1 (~erwan@www.thiscow.com) |
| 13:33.14 | Emrah | Anyone know why the callerid is changed by the IAX Account when I make a connection between two asterisk servers? |
| 13:33.50 | Emrah | thieumS: Quoi comme provider? |
| 13:33.57 | multrix | thieumS: ca m'interesse a titre perso, j'suis en train de voir pour chez moi :) vous faites les particuliers ? t ip centrex koi |
| 13:34.19 | libpcp | what does it mean by this --> exten => s,1,GotoIf($[${LEN(${ARG1})} = 10]?2:4) |
| 13:34.49 | junky[work] | if length of arg1 = 10 goto 2, else goto 4 |
| 13:34.53 | thieumS | multrix: on fait pas les particuliers :( |
| 13:35.02 | thieumS | ca demande une organisation disons ... différente |
| 13:35.12 | thieumS | Emrah: on est pas connu, ça te dira rien |
| 13:35.15 | multrix | thieumS: j'crois que j'vé prendre wengo de 9 telecom ! |
| 13:35.18 | jetscreamer | no space needed between the goto and the lf? (i know nothing) |
| 13:35.26 | Emrah | (C'est à dire?) |
| 13:35.34 | pashah | libpcp: http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20GotoIf#comments |
| 13:35.53 | multrix | http://www.wengo.fr/ |
| 13:36.36 | Emrah | thieumS: Comment s'appelle ton provider? |
| 13:36.53 | jpmcallister | I'm trying to setup as IVR system. I play a menu message with the Background command. If a press a digit before the message stops, it works, But if I wait to the message finish, the line hungs up. Anybody knows what could be wrong? |
| 13:36.57 | thieumS | Emrah: ca te dira rien, tu peux me croire |
| 13:37.20 | thieumS | multrix: leur site est hs on dirait |
| 13:37.46 | Emrah | Anyone know why when a call is made from server-one to server-2, the callerid of the person calling from server-1 is replaced by the IAX account name? (server-1 is connected with iax to server-2) |
| 13:38.33 | pashah | jpmcallister: post ur ivr on pastebin |
| 13:38.55 | *** join/#asterisk JohnJacob (~JohnJacob@pcp619824pcs.mainf01.in.comcast.net) |
| 13:39.20 | Emrah | thieumS: Ton provider a une adresse internet, probablement :) si c'est VoIP |
| 13:40.39 | pashah | Emrah: it is not replaced it is changed somewhat u still can figure out whos calling |
| 13:41.47 | Emrah | I don't understand you pashah |
| 13:42.42 | mac_7 | I would like to get some suggestions for VoIP-Soft-Phones under linux |
| 13:42.58 | mac_7 | kphone & linphone are known |
| 13:44.42 | *** join/#asterisk Casper_UA (~casper@ragu.bestnet.kharkov.ua) |
| 13:46.29 | *** join/#asterisk implicit (~implicit@ip68-5-148-1.oc.oc.cox.net) |
| 13:46.32 | *** join/#asterisk iguy (~iguy@dsl093-197-234.mke1.dsl.speakeasy.net) |
| 13:46.37 | *** join/#asterisk postel (~canonical@host217-42-82-130.range217-42.btcentralplus.com) |
| 13:47.04 | libpcp | okay thanks guys |
| 13:47.16 | *** join/#asterisk porche (~a@dsl81-215-21652.adsl.ttnet.net.tr) |
| 13:47.33 | porche | hi |
| 13:52.19 | implicit | hi |
| 13:53.39 | Emrah | pashah: ? |
| 13:53.51 | thieumS | Emrah: non |
| 13:54.16 | thieumS | on a fait que de la terminaisons pour les opérateurs |
| 13:54.27 | thieumS | on a pas de site internet |
| 13:56.12 | *** join/#asterisk fa (faceoff@devel.acdbddh.eu.org) |
| 13:56.16 | fa | elou |
| 13:57.50 | fa | Jan 31 14:54:32 ERROR[13613]: chan_zap.c:10063 setup_zap: Unable to register channel '1-2' |
| 13:58.07 | jpmcallister | pashah: http://pastebin.ca/5029 |
| 13:58.51 | fa | what module i must load |
| 13:58.58 | implicit | goodnight |
| 13:59.03 | implicit | well its almost time to wake up |
| 13:59.08 | implicit | i guess iahave an hour :) |
| 14:00.34 | ManxPower | fa, The list of modules and what cards they are for is listed in the Zaptel README file. Read it. |
| 14:02.02 | jpmcallister | pasha: If I don't press any digit while the background command of the line 012 is being executed, I get these erros on * console: |
| 14:02.06 | jpmcallister | <PROTECTED> |
| 14:02.06 | jpmcallister | <PROTECTED> |
| 14:02.28 | netsurfer | anyone seen any php scripts for * config files from a mysql database? |
| 14:03.03 | Emrah | Anyone can help me with my strange problem with the callerid with IAX? |
| 14:03.13 | tzanger | Emrah: explain the strange problem |
| 14:03.29 | bjohnson | just to clarify (becuse I found a 2002 thread from someone saying they were adding this), it is not currently possible to configure voicemail to use ldap in any way |
| 14:03.38 | porche | q: how can I know that the card is working? |
| 14:03.44 | porche | X100P |
| 14:03.49 | fa | ManxPower it was gozap.go |
| 14:03.57 | bjohnson | porche: it answers, it dials |
| 14:04.09 | porche | emrah, what kind of connectivity? |
| 14:04.14 | ManxPower | what is a gozap.go? |
| 14:04.23 | porche | bjohnson, ehu tnx, asterisk errors with |
| 14:04.24 | bjohnson | a cheer |
| 14:04.28 | porche | Unable to specify channel |
| 14:04.29 | pashah | jpmcallister: for example add exten => s,9,Goto(s,6) to ur [ura] |
| 14:04.53 | bjohnson | ManxPower: <- a male "cheerleader" |
| 14:04.58 | porche | modprobe zaptel; modprobe wcfxo works |
| 14:05.31 | bjohnson | dmesg says that it "Found" a card? |
| 14:05.42 | bjohnson | or /var/log/messages |
| 14:06.00 | ManxPower | jpmcallister, That's weird. It should complain about no "t" extension found. |
| 14:06.41 | porche | Found a Wildcard FXO: Wildcard X101P |
| 14:06.43 | porche | yep |
| 14:07.05 | jpmcallister | ManxPower: but there is a t extension |
| 14:07.22 | jpmcallister | pashah: the same error ocurred |
| 14:07.26 | ManxPower | Then it should run that. |
| 14:07.43 | ManxPower | Since if you don't enter anything when Background running it will go to exten t |
| 14:07.44 | *** join/#asterisk slash^ (~Susan@220-244-239-233-sa-pppoe.tpgi.com.au) |
| 14:07.51 | *** join/#asterisk sjaak538 (~sjaaknabu@bmr-d8e8.mxs.adsl.euronet.nl) |
| 14:07.51 | jpmcallister | ManxPower: I'm dialing to the ura from an iax phone |
| 14:07.58 | pashah | jpmcallister: have u reloaded conf? |
| 14:08.13 | jpmcallister | ManxPower: yes, but it should at least wait for the timeout |
| 14:08.20 | jpmcallister | pashah: yes |
| 14:08.32 | Emrah | porche: Can I write to you in a private window? |
| 14:08.39 | porche | yep |
| 14:08.44 | ManxPower | jpmcallister, I don't really care what is dialing what. It's not an issue when dealing with Background, etc. Yes it should. If it's not then you have some OTHER issue, like non-consecutive priorities. |
| 14:09.17 | slash^ | hi guys, how reliable is sending a fax over a sip provider and recieveing one... ? |
| 14:09.26 | jpmcallister | ManxPower: my conf is here http://pastebin.ca/5029 |
| 14:09.56 | jpmcallister | ManxPower: The error ocurr afeter line 012 |
| 14:11.17 | ManxPower | Does s,6,Background(esc-menu-principal) ever play? |
| 14:11.21 | sjaak538 | Hello voiptech's how would voipjet make this IAX2/voipjet/1 or IAX2/voipjet/7 how to balance this. |
| 14:11.37 | jpmcallister | ManxPower: yes. And if I press any digit it works |
| 14:11.43 | sjaak538 | they have many servers but i connect to only one |
| 14:12.06 | ManxPower | That was not my question. My question was does the sound file esc-menu-principal play? |
| 14:12.09 | *** join/#asterisk Jas_Williams (~Jason@host217-43-52-65.range217-43.btcentralplus.com) |
| 14:12.27 | jpmcallister | MaxPower: yes, it play |
| 14:12.42 | fa | who is using cdr with postgresql ? |
| 14:12.49 | jpmcallister | ManxPower: I suspected of the iax phone because of that error that appear on * console, right after the s,6,background : |
| 14:12.51 | jpmcallister | == Auto fallthrough, channel 'IAX2/2001@2001/2' status is 'UNKNOWN' |
| 14:13.13 | ManxPower | jpmcallister, What's BEFORE THAT? |
| 14:13.31 | *** join/#asterisk benjr (~benjr@benjr.elo.com.br) |
| 14:13.41 | ManxPower | Seeing extensions.conf in this case is not all that useful without also seeing a pastebin of the CLI output of a sample call. |
| 14:13.45 | porche | question |
| 14:13.57 | jpmcallister | ManxPower: I'll pastebin |
| 14:13.59 | porche | what does unconfigured mean for a x101P mean? |
| 14:14.03 | ManxPower | I do not know if you can use include => in the way tou are using it. |
| 14:14.13 | ManxPower | porche, It means it's not configured. |
| 14:14.22 | *** join/#asterisk mAsH` (~mAsH@host46-29.pool8173.interbusiness.it) |
| 14:14.34 | porche | ok, grt, how can I configure? |
| 14:14.48 | benjr | hi, everybody. I am using ASTERISK + MYSQL for SIPFRIENDS very well. Is there a way to use encrypted passwords in the database? |
| 14:14.50 | jpmcallister | ManxPower: http://pastebin.ca/5031 |
| 14:15.08 | ManxPower | porche, You load the wcfxo kernel driver, you set up /etc/zapata.conf and /etc/asterisk/zaptel.conf. |
| 14:15.17 | bjohnson | porche: after I got the "Found" message .. I had no issues with my X100P |
| 14:15.41 | porche | I did them all, probably kernel modules did not load ok |
| 14:15.46 | porche | let's check it |
| 14:15.56 | slash^ | anyone tried sending a fax over a sip provider and recieveing one... ? |
| 14:16.05 | cereal_ | Jas_Williams hi you re back ? |
| 14:16.10 | tzanger | slash^: won't work |
| 14:16.15 | tzanger | slash^: unless you're VERY lucky |
| 14:16.21 | slash^ | serious ? |
| 14:16.39 | bjohnson | slash^: info on the wiki |
| 14:16.40 | Jas_Williams | cereal_: Yes I am Back |
| 14:16.48 | bjohnson | slash^: also search the mailing list archives |
| 14:16.50 | slash^ | i have a client who wants to use their voip provider for faxing to sav coin |
| 14:16.52 | cereal_ | ok can we continue to sort this ISDN shit ? |
| 14:16.55 | slash^ | yeah ive been reading it |
| 14:16.57 | bjohnson | Jas_Williams: hi !! |
| 14:17.00 | slash^ | alot of diff opinions |
| 14:17.14 | tzanger | slash^: won't work unless you get t.38 into asterisk or work around it with app_rxfax/app_txfax |
| 14:17.37 | porche | is there a specific thing to be done on /etc/zaptel.conf? |
| 14:17.44 | slash^ | but the txfax and rxfax is using a zap channel still yeah ? |
| 14:17.46 | bjohnson | slash^: sounds like a lot of screweing around to get it to work SOME of the time |
| 14:17.50 | tzanger | slash^: yes |
| 14:17.54 | slash^ | not going over a provider :-/ |
| 14:17.55 | tzanger | there is no way around it without t.38 |
| 14:18.05 | ManxPower | jpmcallister, I have no more suggestions |
| 14:18.06 | *** join/#asterisk mtqh (~chatzilla@dsl093-001-038.det1.dsl.speakeasy.net) |
| 14:18.18 | jpmcallister | ManxPower: tanx anyway :) |
| 14:18.20 | slash^ | and t.28 is commerical ? |
| 14:18.55 | tzanger | slash^: t.38 and no -- there's just not support for it in * at this time |
| 14:18.56 | ManxPower | slash^, Asterisk does not support T.38 |
| 14:19.23 | implicit | slash^: someone is working on support right now, but they are not sure if they will make it public they have said |
| 14:19.24 | cereal_ | so still no way to place a outgoing call with ISDN saomeone had the same issue one time ? |
| 14:19.35 | slash^ | hmm |
| 14:19.37 | slash^ | that is sad |
| 14:19.56 | ManxPower | slash^, When can we expect you to release a T.38 driver for Asterisk? |
| 14:19.56 | slash^ | so pstn for faxing i guess |
| 14:20.06 | `Sauron | Anyone know if Theo something-greek hangs out in here? |
| 14:20.07 | tzanger | yup |
| 14:20.12 | slash^ | lol |
| 14:20.16 | tzanger | ManxPower: quit harshing his buzz :-) |
| 14:20.16 | Jas_Williams | cereal_: Did you get adebug trace ? |
| 14:20.21 | ManxPower | slash^, Apprently nobody with the correct skills cares enough to write T.38 for Asterisk |
| 14:20.31 | fa | I have problem.. i am calling to somebody from firefly by asterisk to cellular phone |
| 14:20.32 | *** join/#asterisk mcisse_ (~mcisse@ARennes-303-1-3-177.w217-128.abo.wanadoo.fr) |
| 14:20.32 | cereal_ | Jas_Williams dont move i make you one |
| 14:20.36 | implicit | ManxPower: cept for $$ |
| 14:20.44 | fa | and when sombody dont accept that calls.. i don't have hangup in firefly |
| 14:20.45 | slash^ | i find that interesting considering all business fax |
| 14:20.55 | benjr | Is there a way to use encrypted passwords in MYSQL SIPFRIENDS table? |
| 14:21.08 | multrix | someone has a sip server who has calls free to poland ? |
| 14:21.46 | ManxPower | I didn't know Poland had any local calling. |
| 14:21.59 | ManxPower | Hell, I didn't know FRANCE has free local calling. |
| 14:22.07 | *** join/#asterisk RoyK (~roy@dsl-40-122.kunde.brednett.no) |
| 14:22.18 | multrix | lol |
| 14:22.35 | `Sauron | multrix: Broadvoice.com offers free calls to poland if you sign up for the right account package... |
| 14:27.11 | fa | anybody use psotgres to sip/iax and extasions information? |
| 14:27.34 | porche | where can I find a zapata.conf example for x101P? |
| 14:27.45 | postel | psotgres is obsolete, use postgres |
| 14:27.53 | `Sauron | fa: I tried last night |
| 14:27.55 | fa | postel i mena postgresql |
| 14:28.01 | fa | `Sauron and what effect? |
| 14:28.06 | `Sauron | fa: ast_data needs a fair bit of work, still. |
| 14:28.36 | ManxPower | fa, I don't use any databases with Asterisk |
| 14:28.53 | `Sauron | It looks like it has a chance to work, but I had problems doing extension lookups properly |
| 14:29.13 | `Sauron | That's not a pgsql-specific problem, but it seemed like it was a * issue |
| 14:29.28 | fa | `Sauron a what about mysql.. maybe i must uise mysql |
| 14:29.57 | `Sauron | Nah, like I said - the problem wasn't the DB-specific code |
| 14:30.04 | humblast | This might be a "little" off topic but I hope you bear with me, it is a question about asterisk's codecs: There are some header files in the codecs folder who's names ends with _ex.h ... when I take a look at them I am dumbfounded, I have no clue as to what they are about, does anybody know? |
| 14:30.25 | fa | `Sauron only extasions.. IAX and SIP accounts workign correct? |
| 14:30.34 | `Sauron | fa: seems like it |
| 14:31.09 | postel | humblast: .h are header files, you need them when compiling |
| 14:31.37 | *** join/#asterisk mutilator (~animenodv@65.111.201.79) |
| 14:31.40 | mutilator | FYI |
| 14:31.44 | mutilator | voip in iraq doesn't work |
| 14:31.50 | mutilator | :P |
| 14:31.53 | postel | humblast: no they are NOT configs for your * setup, leave them alone |
| 14:32.04 | ManxPower | mutilator, They set up an anti-voip field? |
| 14:32.15 | mutilator | yea |
| 14:32.17 | mutilator | called latency |
| 14:32.17 | mutilator | <PROTECTED> |
| 14:32.17 | mutilator | Approximate round trip times in milli-seconds: |
| 14:32.17 | mutilator | <PROTECTED> |
| 14:32.19 | ManxPower | Celever. |
| 14:32.37 | ManxPower | Wish I had an anti-cell phone field generator. |
| 14:32.53 | mutilator | had a few people go over there with ata's |
| 14:33.00 | mutilator | and so far none have had any luck |
| 14:33.36 | ManxPower | I must admit that someone bringing an ATA to Iraq is someone that should be referenced in the dictionary in the definition of "optimist" |
| 14:33.40 | vaewyn | ManxPower: try a HERF gun... works for several of those little annoyances |
| 14:34.00 | mutilator | yea, but if it did work it'de be great |
| 14:34.13 | *** join/#asterisk fantomax1 (~fanto@81.208.114.250) |
| 14:34.17 | fantomax1 | hi all |
| 14:34.22 | ManxPower | vaewyn, You have no idea how much I want one of those. Mostly to deal with Urban Youth sharing their music when in their car. |
| 14:34.22 | mutilator | gotta be an optimist if you're goin to iraq anyway |
| 14:34.31 | mutilator | hopin ya don't get ya arm blown off |
| 14:34.39 | fantomax1 | is there anyone that use SIPP for producing SIP calls ? |
| 14:34.43 | humblast | postel: Guess the question might better fit in the asterisk-dev channel... thanks for the reply |
| 14:34.52 | benjr | Is there a way to use encrypted passwords in MYSQL SIPFRIENDS table? anyone, please |
| 14:35.11 | vaewyn | ManxPower: :} |
| 14:36.26 | `Sauron | benjr: Probably. And we probably don't know how to. |
| 14:36.40 | *** join/#asterisk cbachman (~cbachman@129.105.7.250) |
| 14:36.43 | ManxPower | vaewyn, Where I live now, the Suburban Youth are usually playing country music and have a gunrack on the back of their pickup truck. Much less loud. |
| 14:37.07 | ManxPower | But at least THEIR guns are in plain sight. 8-) |
| 14:37.44 | vaewyn | ManxPower: I trust the viewable ones way more |
| 14:37.50 | ManxPower | vaewyn, Me too. |
| 14:38.13 | `Sauron | Don't know |
| 14:38.44 | netsurfer | benjr - what about MD5 passwords ? |
| 14:39.26 | *** join/#asterisk JohnJacob (~JohnJacob@pcp619824pcs.mainf01.in.comcast.net) |
| 14:41.50 | *** join/#asterisk eKo1 (~bernd@63.245.57.70) |
| 14:43.53 | *** join/#asterisk keith778 (~keith778@ool-4355f47e.dyn.optonline.net) |
| 14:44.27 | keith778 | Anyone here good with AGI?? I am having a problem receiving the environment variables |
| 14:44.29 | fa | who use realtime? |
| 14:45.33 | ManxPower | keith778, Only recent CVS-HEAD allows you to get autocreated variables from within AGI. |
| 14:46.51 | keith778 | Oh, ok. This may be my problem. I am running CVS 1/21/05 Does this CVS have an issue? |
| 14:47.41 | netsurfer | still getting that error with sipphone.com.. "please check your number,, the user you are trying to reach is unknown" |
| 14:47.55 | *** join/#asterisk PakiPenguin (~info@202.176.254.1) |
| 14:48.07 | *** join/#asterisk carbon60 (~adam@Quebec-HSE-ppp230772.qc.sympatico.ca) |
| 14:48.10 | netsurfer | fa - i got realtime working |
| 14:48.12 | carbon60 | Morning all. |
| 14:48.14 | keith778 | I am trying to capture the env variables such as "agi_callerid:" My script works fine with a test file sent to stdin. When it tries to read the env variable from * it doesn't work |
| 14:48.16 | ManxPower | keith778, I don't know. The workaround for a very long time has been something like SetVar(MY_CAUSECODE=${CAUSECODE}) before running the AGI. Of course the common variables are available when you read your STDIN when starting your AGI. |
| 14:48.37 | ManxPower | keith778, Are you using Perl? |
| 14:48.40 | carbon60 | Does anyone know the sox incantation to convert from "WAV" (GSM compressed) to normal "wav"? |
| 14:48.43 | keith778 | No Python |
| 14:48.52 | ManxPower | keith778, You are on your own them. |
| 14:49.00 | `Sauron | carbon60: untoast |
| 14:49.04 | ManxPower | carbon60, You mean like the billion examples on the Wiki and on the mailing list archives. |
| 14:49.10 | carbon60 | Uh oh. |
| 14:49.26 | carbon60 | Ignore me then! I'm off to the Wiki. |
| 14:50.57 | *** part/#asterisk keith778 (~keith778@ool-4355f47e.dyn.optonline.net) |
| 14:52.00 | *** join/#asterisk SuPrSluG (~SuPrSluG@pool-70-16-40-16.buff.east.verizon.net) |
| 14:52.28 | SuPrSluG | hello |
| 14:52.41 | SuPrSluG | small extensions problem. if i enable my ITSP it goes to that w/out using dundi. any sample dialplans for using dundi? |
| 14:54.18 | carbon60 | ManxPower: I can find lots of examples for doing the reverse... Have a URL? |
| 14:54.32 | multrix | somebody know a good programm Php or other to have a web interface to asterisk ? |
| 14:54.59 | carbon60 | Also, I believe I'm experiencing the same issue as others on the list where the compressed files are mush software than the uncompressed versions. Did anyone find a solution to this? |
| 14:55.10 | ManxPower | carbon60, No. but if you see examples to do the reverse it should not be hard to find a way to figure it out. |
| 14:55.27 | ManxPower | mush software?? |
| 14:55.33 | ManxPower | Don't cook it for so long. |
| 14:56.21 | *** join/#asterisk olivier_ (~olivier_@AMarseille-152-1-1-32.w82-127.abo.wanadoo.fr) |
| 14:56.51 | *** join/#asterisk keith778 (~keith778@ool-4355f47e.dyn.optonline.net) |
| 14:57.12 | *** join/#asterisk McKillroy (~mckillroy@L0824P27.dipool.highway.telekom.at) |
| 14:57.12 | keith778 | Ok, I have a general AGI question...not related to python |
| 14:57.13 | McKillroy | Hello ! |
| 14:57.35 | McKillroy | How far is an * Bluetooth hands free channel ? |
| 14:57.36 | ManxPower | keith778, then I refer you to asterisk-perk |
| 14:57.47 | ManxPower | asterisk-perl too |
| 14:58.05 | keith778 | ManxPower: Ok, thanks |
| 14:58.19 | McKillroy | And: is the Dock-N-Talk Cell phone adapter a possibility to make a GSM gateway ? |
| 14:59.46 | McKillroy | Link: https://www.phonelabs.com/prd05.asp |
| 15:00.16 | *** join/#asterisk felipex (~dsfdsf@host179-130.pool8172.interbusiness.it) |
| 15:00.28 | ManxPower | McKillroy, With most of those devices Asterisk doens't know when the far end hangs up. |
| 15:01.35 | McKillroy | So - could it be used at least for outgoing calls maybe ? |
| 15:01.39 | *** join/#asterisk Jas_Williams (~Jason@host217-43-52-65.range217-43.btcentralplus.com) |
| 15:02.32 | jpmcallister | what is a good linux sip software phone? |
| 15:02.36 | McKillroy | Better would probably be a Bluetooth Channel ... |
| 15:02.50 | eKo1 | What's the syntax for droping the first three digits (for example) of a number? |
| 15:03.07 | vaewyn | eKo1: $VARIABLE:3} |
| 15:03.13 | vaewyn | +{ |
| 15:03.14 | olivier_ | ${EXTEN:3} |
| 15:03.22 | eKo1 | Thank you. |
| 15:04.39 | fa | why can i get that error |
| 15:04.41 | fa | <PROTECTED> |
| 15:04.41 | fa | Jan 31 16:04:08 NOTICE[31372]: app_dial.c:762 dial_exec: Unable to create channel of type 'IAX2' |
| 15:04.44 | fa | ? |
| 15:05.20 | Jas_Williams | IAX2/inezk is not configured/registered or rejects your call |
| 15:05.34 | fa | no. |
| 15:05.41 | fa | by inezk i Can call ouy.. |
| 15:05.46 | fa | but not accept calls |
| 15:06.05 | fa | That coudl by becuase inezk - friefly hac configured only alaw - as protocol |
| 15:06.07 | fa | copression |
| 15:06.54 | Jas_Williams | fa do a iax2 show peers |
| 15:07.03 | Jas_Williams | in the CLI |
| 15:07.14 | fa | Name/Username Host Mask Port Status |
| 15:07.14 | fa | inezk1/inezk1 (Unspecified) (D) 255.255.255.255 0 Unmonitored |
| 15:07.14 | fa | inezk/inezk (Unspecified) (D) 255.255.255.255 0 Unmonitored |
| 15:07.31 | Jas_Williams | fa the phones are not registered to * |
| 15:07.53 | fa | Jas_Williams but I am .. I can make a call from that firefly.. |
| 15:08.09 | fa | <PROTECTED> |
| 15:08.12 | fa | <PROTECTED> |
| 15:08.23 | Jas_Williams | fa you do not need to be registered to originate a call |
| 15:08.31 | fa | ohh.. so how to register? |
| 15:08.40 | fa | a i know |
| 15:08.46 | fa | i have checked disable registration |
| 15:08.58 | fa | <PROTECTED> |
| 15:08.58 | fa | ;] |
| 15:09.08 | Jas_Williams | Now it should work :-) |
| 15:11.29 | *** join/#asterisk jaiger (~jaiger@fire.innovationsw.com) |
| 15:12.00 | *** join/#asterisk stickynomore (~jeff@nsc66.147.11-46.newsouth.net) |
| 15:13.56 | *** join/#asterisk randu (~randu@dsl-pppoe-pool1-37.d02.ncshlt.infoave.net) |
| 15:14.49 | *** join/#asterisk Ubuz (~momo@HFA62-0-190-120.bb.netvision.net.il) |
| 15:16.36 | multrix | somebody knows a good advanced tutorial about asterisk ? |
| 15:16.46 | *** part/#asterisk keith778 (~keith778@ool-4355f47e.dyn.optonline.net) |
| 15:17.11 | fa | multrix hm asteriskdocs.org or sth liek that.. but is not more advanced |
| 15:17.58 | eKo1 | Hmm...for some reasone sip reload isn't working. |
| 15:19.19 | fa | eKo1 i use IAX2.. but i don't have a good example to realtiem reload |
| 15:19.40 | freat | multrix: have you been to http://www.voip-info.org/ ?? |
| 15:20.23 | *** join/#asterisk mogorman (~mogorman@dhcp-162.digium.com) |
| 15:20.31 | jaiger | multrix: the best advanced tutorial is dig in and get your hands dirty |
| 15:21.09 | olivier_ | ~docs |
| 15:21.09 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 15:22.06 | tzanger | ~seen jtodd |
| 15:22.11 | jbot | jtodd <~jtodd@c-24-22-6-31.client.comcast.net> was last seen on IRC in channel #asterisk, 3d 8h 48m 43s ago, saying: 'blitzrage: Nope. Toronto is waaaay too cold.'. |
| 15:22.31 | tzanger | does anyone (besides jtodd) know much about the Tellabs echo cancellers? |
| 15:25.02 | jaiger | tzanger: any day now I'm going to get one up and running |
| 15:25.14 | jaiger | tzanger: but I don't have any experience yet, other than reading |
| 15:26.50 | mAsH` | who can help me? |
| 15:27.05 | mAsH` | i have a problem with compiling * |
| 15:29.23 | tzanger | mAsH`: nobody can help you if you aren't giving us any more info |
| 15:30.09 | *** join/#asterisk mhnoyes (~mhnoyes@user-2ivfndp.dialup.mindspring.com) |
| 15:30.16 | mAsH` | make[1]: *** [pbx_gtkconsole.so] Error 1 |
| 15:30.16 | mAsH` | make[1]: Leaving directory `/root/asterisk-1.0.5/pbx' |
| 15:30.16 | mAsH` | make: *** [subdirs] Error 1 |
| 15:30.16 | mAsH` | root@lite:~/asterisk-1.0.5# |
| 15:30.22 | *** part/#asterisk djin (~marius@62.58.40.196) |
| 15:31.33 | olivier_ | is there any other line before "make[1]: *** [pbx_gtkconsole.so] Error 1" ? |
| 15:31.36 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-181-141.dsl.scarlet.be) |
| 15:31.46 | ManxPower | mAsH`, something is wrong. things should not fail when gtkconsole fails to build. |
| 15:31.50 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 15:32.07 | Zeeek | good ${LocalTimeOf The Day} |
| 15:32.19 | olivier_ | :) Hi zeek :) |
| 15:32.22 | Zeeek | anyone know if e164.org is having problems? |
| 15:32.37 | Zeeek | 'lo olivier |
| 15:33.16 | eKo1 | Argh. Why can't all call terminators agree on an international calling prefix? |
| 15:33.21 | mAsH` | olivier_: those |
| 15:33.21 | mAsH` | /usr/lib/gcc-lib/i486-slackware-linux/3.3.4/../../../../i486-slackware-linux/bin/ld: cannot find -lXext |
| 15:33.21 | mAsH` | collect2: ld returned 1 exit status |
| 15:34.11 | eKo1 | It would be nice if * had regex support... |
| 15:34.13 | Zeeek | eKo1 they need to agree about a lot of other shit first, like Iraq! |
| 15:34.25 | ManxPower | mAsH`, Those should not be critical errors. |
| 15:35.40 | Zeeek | Manx do you know why there would be an accumulation of register attempts in sip channels? |
| 15:36.18 | Zeeek | I know one provider drops out of reg several times a day but why would the reg attelmptsz leave dead channels? |
| 15:37.27 | *** join/#asterisk daveg (~root@80.46.98.226) |
| 15:39.04 | *** join/#asterisk daveg (~daveg@80.46.98.226) |
| 15:41.14 | eKo1 | Oh! There are regular expressions. My bad. |
| 15:41.30 | Zeeek | regular expressions in what? |
| 15:41.52 | *** join/#asterisk calvinhp (~calvinhp@rrcs-24-123-25-236.central.biz.rr.com) |
| 15:42.12 | freat | hey isn't stable 1.2 due out before too long? |
| 15:42.23 | freat | realtime? |
| 15:42.59 | ManxPower | freat, Details were discussed on the mailing list, even with a link to Mark's talk about release timeframes. |
| 15:43.17 | ManxPower | I guess you are not on the mailing lists, huh, freat? |
| 15:43.18 | *** join/#asterisk JohnJacob (~JohnJacob@pcp0011543956pcs.mainf01.in.comcast.net) |
| 15:43.19 | freat | ManxPower: oh thanks! missed the thread |
| 15:43.33 | freat | ManxPower: no I'm on them both, but don't have time to read everything |
| 15:43.40 | ManxPower | freat, Might have been on -dev |
| 15:43.42 | fa | is available a pgsql modula like MYSQL ? to make a inser or selects from extensions.conf ? |
| 15:43.49 | freat | cool thx |
| 15:44.08 | fa | like that |
| 15:44.10 | fa | exten => s,2,MYSQL(Connect connid mysqls.esh.pl webuser mojephpy1 asterisk) |
| 15:44.10 | fa | exten => s,3,MYSQL(Query resultid ${connid} SELECT\ `imie`\ FROM\ `user`\ WHERE\ `numer` like \'\%${CALLERID}\%\') |
| 15:44.13 | fa | exten => s,4,MYSQL(Fetch foundRow ${resultid} imie) |
| 15:45.10 | fa | (only read) |
| 15:47.39 | Zeeek | NAT for Dummies, eh? |
| 15:47.46 | ManxPower | Zeeek, Yeah. |
| 15:47.56 | ManxPower | It's prolly really called NAT for Managers. |
| 15:48.14 | Zeeek | I thought it was called the asterisk mailing list |
| 15:50.44 | *** join/#asterisk amir (~amir@shield.guindehi.ch) |
| 15:51.11 | vaewyn | ManxPower: same thing |
| 15:51.13 | vaewyn | :} |
| 15:52.33 | *** join/#asterisk TheEmperor (TheEmperor@218.111.51.101) |
| 15:52.52 | fa | how to install something like MYSQL but for postgres? |
| 15:54.35 | *** join/#asterisk bprice20 (~brandon@Dynamic-216.120.224.151.hrnoc.net) |
| 15:55.49 | bprice20 | hey w/ realtime what the syntax for extconfig.conf when i want it to use a mysql database thats on another host |
| 15:56.25 | Zeeek | vaewyn - what a good sport you are :) |
| 15:56.31 | *** join/#asterisk hmmhesays (~hmmhesays@66.173.103.108) |
| 15:56.43 | Zeeek | always a positive word |
| 15:57.10 | Zeeek | olivier_ ? |
| 15:57.41 | Zeeek | Yodid I do that? |
| 15:58.17 | Zeeek | I have a quick question about dégroupage - I think olivier_ you may know about that? |
| 15:59.01 | *** join/#asterisk file[laptop] (~file_lapt@mctn1-142166196242.nb.aliant.net) |
| 15:59.07 | vaewyn | Zeeek: I'm an optomist... but a realist also ;P |
| 15:59.21 | Zeeek | I'm a cynic but an optimist in some areas |
| 15:59.59 | nestAr | i'm a realist... which most people mistake for cynicism. |
| 16:00.07 | Zeeek | I think, for example, that when the earth has been struck by a huge meteorite we'll have solved most of our problems, but it'll be too late |
| 16:00.18 | ellvis | pesimist is just good informed optimist :) |
| 16:00.28 | Zeeek | ellvis is alive |
| 16:00.35 | ellvis | hi Zeeek |
| 16:00.39 | ellvis | i am surviving |
| 16:00.42 | vaewyn | yep... he just went home |
| 16:01.08 | Zeeek | I always laughed when I saw the pseudo on a grc tech newsgroup: BloatedElvis |
| 16:01.13 | ellvis | on last concert i broke my leg and people mistaken it |
| 16:01.20 | ellvis | they believe i died :) |
| 16:01.34 | *** join/#asterisk djMax (~djMax@artsalliancelabs.com) |
| 16:01.39 | blitzrage | ~seen zx81 |
| 16:01.40 | jbot | zx81 <~ZX81@222-152-158-141.jetstream.xtra.co.nz> was last seen on IRC in channel #asterisk, 1d 14h 24m 36s ago, saying: 'yeah'. |
| 16:01.46 | Zeeek | Anyone reading trhz National Enquirer knows not only you are alive but that you are running the US govt! |
| 16:02.25 | Zeeek | ellvis has left the channel... ellvis has left the channel |
| 16:02.43 | djMax | ok, so if I want to bang my head against * h.323 again should I go on inaccess' version or the other one? |
| 16:02.44 | Zeeek | speaking of NAT and double NAT |
| 16:02.59 | ellvis | Zeeek, should i stay or should i go? :) |
| 16:03.08 | Zeeek | heh good one! |
| 16:03.44 | Zeeek | I can't even remember where that "elvis has left the building" came from |
| 16:03.55 | gambolputty | probably some concert |
| 16:04.01 | Zeeek | something to do with Gracelabnd tours |
| 16:04.13 | Zeeek | no when folks are touring his home |
| 16:04.22 | Zeeek | when he was alive and living there |
| 16:04.27 | ellvis | yes |
| 16:04.31 | ellvis | that's it |
| 16:04.32 | ManxPower | djMax, Generally go with the chan_h323 included with Asterisk |
| 16:04.53 | Zeeek | http://www.straightdope.com/columns/021227.html |
| 16:04.56 | ellvis | Zeeek, but i broke my leg for real anyway. last friday on concert... :( |
| 16:05.03 | djMax | ok gud, thanks. |
| 16:05.20 | djMax | (trying to get voice connection to a polycom viewstation) |
| 16:05.35 | Zeeek | I was wrong |
| 16:05.38 | Zeeek | After Elvis had given his final encore and left the stage, the crowd headed for the exits, even though many other Hayride acts were still waiting to perform. Logan took the microphone and pleaded with Elvis's fans to return to their seats: "Please, young people . . . Elvis has left the building. He has gotten in his car and driven away. . . . Please take your seats." |
| 16:07.47 | tangel | is there any free local PSTN calling networks anymore? |
| 16:08.00 | tangel | like inphonex used to allow free PSTN calls to certain area codes even for their free accounts |
| 16:08.35 | WildPikachu[BAR] | cool |
| 16:09.00 | WildPikachu[BAR] | i got 2 ISDN cards today from diff places... came back and they were exact same model cards ... heh |
| 16:09.04 | Zeeek | tangel you could try bellster.net |
| 16:09.13 | WildPikachu[BAR] | i'm now going to try setup asterisk + capi 2.0 |
| 16:09.21 | ManxPower | Most places in the USA and canada provide free local calling. |
| 16:09.45 | tangel | zeek, that's awesome.. PPTP PSTN network over IP |
| 16:09.52 | tangel | i'm all for that sort of thing |
| 16:09.55 | WildPikachu[BAR] | ManxPower, heh... wouldnt i just LOVE a IAX termination point there!!!! |
| 16:10.18 | eKo1 | Anybody good with regex here? |
| 16:10.27 | WildPikachu[BAR] | eKo1, what u need? |
| 16:10.57 | *** join/#asterisk fenlander (~irc@82.152.81.57) |
| 16:11.04 | eKo1 | \(1.\+\)|\(011\(.\+\)\) <--- Should this return the match of a number beginning with 011? |
| 16:12.21 | WildPikachu[BAR] | \(011\(.+\)\) |
| 16:12.26 | fa | what about that postgres>? |
| 16:12.42 | WildPikachu[BAR] | hrmmm |
| 16:12.50 | *** join/#asterisk djin (~marius@gridfox.xs4all.nl) |
| 16:13.01 | WildPikachu[BAR] | depends if you want the + a char or to specify that there must be at least one . (char) |
| 16:13.12 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 16:13.19 | WildPikachu[BAR] | eKo1, example of a number? |
| 16:13.19 | eKo1 | It matches. The only thing is, it only returns the number of characters matched. |
| 16:13.24 | PakiPenguin | Hello Everyone, what ports do i need open/forwarded to my system for iaxphone ( any iax client ) to work with *? |
| 16:13.33 | tangel | how can one manage dynamic call routing? |
| 16:13.38 | eKo1 | 011987654321 |
| 16:13.46 | eKo1 | It should return 987654321 |
| 16:13.53 | tangel | like have a system that queries rates on different VOIP providers and then prefs the routes appropriately? |
| 16:13.55 | eKo1 | But it returns 12. |
| 16:13.55 | WildPikachu[BAR] | what does it return? |
| 16:14.00 | WildPikachu[BAR] | hrmmmmm |
| 16:14.16 | WildPikachu[BAR] | 011\(.+\) |
| 16:14.39 | eKo1 | I tried that and I get the same thing. I'll try it again. |
| 16:14.48 | ManxPower | eKo1, Using perl? |
| 16:15.38 | eKo1 | Yep, I still get 12. |
| 16:15.46 | ManxPower | put the result variable in (). i.e. ($bob) = $fred ~= theregex |
| 16:15.49 | eKo1 | ManxPower: I'm using * |
| 16:16.01 | ManxPower | eKo1, I didn't know Asterisk supported regexs. |
| 16:16.18 | eKo1 | It does, exprs1 : regexp |
| 16:16.25 | fa | where can i get a module for postgresql to * ? |
| 16:16.32 | vaewyn | Wasssaaab....err... wasim! |
| 16:16.43 | tzanger | hahahah |
| 16:17.20 | wasim | :) |
| 16:17.33 | eKo1 | Maybe it's a bug in the regex code. Hmm... |
| 16:17.39 | Zeeek | wasim mon ami - comment cas-tu ? |
| 16:17.45 | Zeeek | errrr vas-tu |
| 16:17.49 | psywar | I'm trying to send a FAX over * |
| 16:17.53 | psywar | I wonder if it will work |
| 16:18.01 | `Sauron | fa: I already told you, ook at voip-info.org |
| 16:18.07 | wasim | Zeeek: bien, we should have good news for you shortly |
| 16:18.26 | Zeeek | $shortly= 128 days; |
| 16:18.56 | wasim | Zeeek: 'tis but the blink of an eye in the greater scheme of things |
| 16:18.58 | Zeeek | but I am not one to spit on good news! |
| 16:19.05 | Zeeek | This is very true, wasim |
| 16:20.09 | wasim | Zeeek: ssshhh! don't tell anyone, but we're waiting for the first shipment to pass through EU customs |
| 16:20.22 | Zeeek | oh? |
| 16:21.04 | wasim | oui, mon ami, is truth ... |
| 16:21.14 | Zeeek | this calls for a premature ej^h^h celebration |
| 16:21.32 | `Sauron | fa, I told you to search voip-info.org for "ast_data" |
| 16:21.54 | Zeeek | google for app_pr0n_call_center |
| 16:22.06 | Zeeek | my latest devel project |
| 16:22.11 | PakiPenguin | Zeeek: would land @ your * |
| 16:22.44 | Zeeek | good evening - you guys are up late, non? |
| 16:23.09 | wasim | Zeeek: non, gmt+5 |
| 16:23.13 | PakiPenguin | 21:23:53 |
| 16:23.16 | fa | `Sauron i search for that |
| 16:23.18 | Makenshi | afternoon, it's 16:23 here |
| 16:23.20 | Zeeek | so that's ${HERE} + 4 |
| 16:23.30 | `Sauron | I just searched, and it's the very first result |
| 16:23.31 | Makenshi | gmt |
| 16:23.40 | `Sauron | Then read the documentation for how to configure it |
| 16:23.41 | Zeeek | Nine thirty at night? Whoa, you folks should be in bed! |
| 16:23.55 | *** join/#asterisk doughecka_ (~dheckaman@doughecka.user) |
| 16:23.56 | wasim | Zeeek: ${THERE} == ${landofgoodcheesebreadandwine} |
| 16:24.08 | Zeeek | not to mention women and song |
| 16:24.09 | wasim | Zeeek: we never got out! |
| 16:24.16 | PakiPenguin | :) |
| 16:24.22 | Zeeek | in order to go out you have to go in! |
| 16:24.25 | wasim | Zeeek: we like our women shorn! |
| 16:24.40 | PakiPenguin | wasim: ${there}==${here} too |
| 16:24.45 | Zeeek | this is getting too weird even for #asterisk |
| 16:25.03 | wasim | PakiPenguin: no, we don't have good bread, wine or cheese here |
| 16:25.14 | wasim | PakiPenguin: passable, but not good |
| 16:25.18 | psywar | things can't be too weird |
| 16:25.25 | mutilator | there a good way to check for null strings? |
| 16:25.27 | mutilator | i use "" |
| 16:25.29 | PakiPenguin | wasim: good for where we live :) |
| 16:25.31 | mutilator | but it just gives me an error |
| 16:26.04 | psywar | I want to set up a "pick your own adventure" game using *. When you call me it will say "an angry dwarf blocks your way. press 1 to fight him, 2 to run past him, and 3 to retreat" |
| 16:26.15 | netsurfer | lmao |
| 16:26.15 | Zeeek | mutilator someone kindly shared this with me |
| 16:26.24 | netsurfer | on a premium rate number no less, psywar |
| 16:26.29 | psywar | good idea |
| 16:26.31 | ManxPower | psywar, Set it up as a front end to getting to tech support. |
| 16:26.32 | netsurfer | :) |
| 16:26.44 | Zeeek | mutilator: GotoIf($[X${CALLERIDNUM} = X]?s,7) |
| 16:26.47 | doughecka_ | who has used the sipura spa-841 phone? |
| 16:26.49 | doughecka_ | does it work ok? |
| 16:26.53 | mutilator | ingenius |
| 16:26.54 | mutilator | heh |
| 16:26.58 | mutilator | th |
| 16:26.59 | mutilator | x |
| 16:27.00 | Zeeek | an old shelll trick |
| 16:27.03 | netsurfer | lmao if u fight the dwarf and win, u get to tech support else u gotta cal back |
| 16:27.04 | mutilator | yea |
| 16:27.24 | Zeeek | I had the worst hotline experience a couple of days ago |
| 16:27.30 | psywar | It will measure your caller's abilitty to adapt to new situations and do creative problem solving. |
| 16:27.46 | netsurfer | lmao |
| 16:27.48 | psywar | Weed out people you really didn't want to ttalk to anyway. |
| 16:28.11 | netsurfer | pipe the noobs to /dev/null |
| 16:28.24 | Zeeek | that's what our isp did: they kept us on hold for exactly 1min01 before saying no one is available. That way we paid $0.50 to them for the call |
| 16:28.38 | doughecka_ | lol |
| 16:28.45 | Zeeek | I emailed and the answer two days later was to call the hotline |
| 16:28.57 | Zeeek | this is when we're disconnected at the office |
| 16:29.15 | doughecka_ | how did you email |
| 16:29.17 | doughecka_ | . |
| 16:29.29 | *** join/#asterisk terrapen (~cjs@fw-01.satx.bikeworld.net) |
| 16:29.34 | doughecka_ | hah, running away! |
| 16:29.35 | doughecka_ | :P |
| 16:29.41 | Zeeek | we have TWO DSL connx since the cable died for three weeks last year |
| 16:29.54 | doughecka_ | ah |
| 16:29.54 | Zeeek | from two diff providers obviously |
| 16:29.55 | *** join/#asterisk zbysio (~zbysio@chello084010031149.chello.pl) |
| 16:30.09 | Zeeek | plus a third at home (10 min on foot) |
| 16:30.13 | doughecka_ | lol |
| 16:30.27 | terrapen | i wonder which is really better |
| 16:30.30 | terrapen | DSL or cable... |
| 16:30.30 | netsurfer | 5 if ur boss makes u run ;) |
| 16:30.41 | Zeeek | except I'm the boss and I always run |
| 16:30.45 | doughecka_ | 1 if a killer bee swarm is behind you |
| 16:30.55 | Zeeek | well... my wife is but she isn't here right now :) |
| 16:31.10 | ManxPower | Mink Louisiana finally got phone service recently. It's one of the last places in the USA to get phone service. |
| 16:31.23 | ManxPower | The state actually had to ORDER the ILEC to install the service to the town. |
| 16:31.23 | netsurfer | wow ur kidding right |
| 16:31.36 | terrapen | heh |
| 16:31.40 | terrapen | read that this morning |
| 16:31.51 | Zeeek | Mink Deville |
| 16:31.52 | netsurfer | I thought my town was behind, we only got DTMF dialling 5 years ago |
| 16:32.19 | terrapen | i wonder if they offer DSL in Mink yet |
| 16:32.39 | terrapen | "When can we get our cable modem? |
| 16:32.51 | doughecka_ | VoIP :D |
| 16:32.59 | fenlander | *CLI> help |
| 16:32.59 | hmmhesays | what's voip? |
| 16:32.59 | fenlander | Segmentation fault |
| 16:33.20 | wasim | hmmhesays: its this new ignition system in cars |
| 16:33.40 | hmmhesays | sweet.... is that better than *points* ? |
| 16:33.47 | eKo1 | voip == very ordinary intelligent people |
| 16:33.59 | wasim | hmmhesays: yep, virtual oil iginition pointless |
| 16:34.35 | hmmhesays | hmm... does it use low resistance wires to screw up everyone's radio signals? |
| 16:35.07 | vaewyn | bwahaha... norhell guy is saying their SIP is better because it "isn't VoIP like asterisk" BWahahahahahaha |
| 16:35.18 | wasim | vaewyn: lol |
| 16:35.24 | fa | `Sauron are you there? |
| 16:35.29 | hmmhesays | nortel guy must have his dunce cap on today |
| 16:35.40 | vaewyn | Now I know there are smart norhell guys... but... this ain't their day :P |
| 16:35.42 | doughecka_ | LOL |
| 16:35.48 | ManxPower | "Asterisk's SIP is better because it's cheaper" |
| 16:36.01 | doughecka_ | "Asterisk's SIP is better because it's not nortel" |
| 16:36.16 | vaewyn | I just said... "well... give it to me for free and let me audit the code and we have a deal" :} |
| 16:36.16 | freat | I think we _finally_ ironed out the problems with our ISP and such. Man, we had an old Netopia router and it was maxxing out too |
| 16:36.50 | freat | the whole site is 'virtual' |
| 16:37.00 | freat | VoIP + Citrix + Videoconferencing |
| 16:37.08 | freat | They all run thin clients |
| 16:37.15 | doughecka_ | do they run vmware? |
| 16:37.21 | freat | I run vmware |
| 16:37.25 | freat | it r0x0rz |
| 16:37.27 | ManxPower | freat, none of the 24 results of your mailing list search for netopia gave you any clue?? |
| 16:37.27 | doughecka_ | woot |
| 16:37.29 | doughecka_ | vmware rox |
| 16:37.31 | djMax | So I'm trying to convince someone that Skype sucks because it can't interface with existing Voip systems well. Am I right? |
| 16:37.34 | freat | hahaha yeah |
| 16:37.42 | doughecka_ | <-- vmware certified :) |
| 16:37.47 | *** join/#asterisk Bicster (~Bicster@bicster.user) |
| 16:37.52 | Bicster | does asterisk work with vonage these days? |
| 16:37.53 | *** join/#asterisk mogorman (~mogorman@dhcp-162.digium.com) |
| 16:37.56 | freat | ManxPower: replaced it with a mikrotik (recommended by mailing list) |
| 16:38.01 | bjohnson | djMax: that's one reason |
| 16:38.09 | ManxPower | Bicster, Yes, with significant limitations |
| 16:38.11 | freat | ManxPower: mikrotik is quite nice |
| 16:38.12 | bjohnson | djMax: another is that they're typically USB |
| 16:38.30 | Bicster | ManxPower, what does that mean? |
| 16:38.33 | bjohnson | Bicster: there are better voip providers .. go read the wiki |
| 16:38.39 | bjohnson | ~docs |
| 16:38.40 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 16:38.40 | Bicster | I already have nufone |
| 16:38.41 | Bicster | works great |
| 16:38.45 | Bicster | but I need a houston line now |
| 16:38.55 | ManxPower | Bicster, Only works with Vonage Softphone Account, which is only an ADDON to their regular service and does not have unlimited calling. |
| 16:39.01 | djMax | bjohnson, why is that bad? |
| 16:39.09 | Bicster | AT&T is merging with SBC so I need new local service |
| 16:39.16 | bjohnson | Bicster: you won't be able to use their hardware. you can get better prices alsewhere |
| 16:39.26 | bjohnson | djMax: needs drivers |
| 16:39.42 | djMax | ah, as opposed to ethernet you mean? |
| 16:39.50 | bjohnson | djMax: yes |
| 16:40.03 | djMax | got it, thx. |
| 16:40.50 | bjohnson | djMax: only question of time until someone decides it isn't worth their time to support the drivers anymore |
| 16:40.54 | Bicster | what do I need to look for on the wiki? |
| 16:41.04 | bjohnson | djMax: eg .. rewrite driver for new versions of OS |
| 16:41.22 | bjohnson | Bicster: you could start with nufone |
| 16:41.30 | Bicster | I already have nufone |
| 16:41.34 | djMax | yeah, and with Skype's attitude they're unlikely to give out the info for someone else to write it |
| 16:41.39 | bjohnson | Bicster: one of the hits will be a page with hundreds of others |
| 16:41.47 | ManxPower | Nufone only provides local DIDs in Michican. |
| 16:42.05 | bjohnson | djMax: exactly .. and limited OS support |
| 16:42.30 | *** join/#asterisk networ (~nobody@datitel.avonet.cz) |
| 16:42.35 | networ | hello |
| 16:42.43 | Bicster | ok, do I need to click on each of the ~50 providers here or can someone suggest a few to check out? |
| 16:42.44 | bjohnson | djMax: equals limit on what you can use it on, how long you can use it, and what you can do with it |
| 16:42.54 | Zeeek | Bicster Voicepulse and ICH |
| 16:42.59 | Bicster | thanks Zeeek |
| 16:43.01 | mutilator | use mine! |
| 16:43.02 | Zeeek | I have tested both |
| 16:43.05 | wasim | Bicster: teleban.af |
| 16:43.09 | Bicster | lol wasim |
| 16:43.11 | bjohnson | Bicster: they all offer different DID areas. Where were you when I was looking for London DIDs? |
| 16:43.15 | networ | can anybody help me how i can configure ast sip and mysql ? |
| 16:43.33 | mutilator | all i have is michigan did's too |
| 16:44.48 | bjohnson | Bicster: one difference among providers is how many concuurent calls can be handled |
| 16:44.49 | Bicster | ICH isn't on the wiki AFAICT |
| 16:45.12 | Bicster | I don't care about that -- I just need a local # for people to call, with 911 service |
| 16:45.14 | bjohnson | fa: post your question here. Don't pm me |
| 16:45.43 | Zeeek | ICH works very well (call quality) with asterisk but I do not have any DID with them |
| 16:46.00 | Zeeek | IconnectHere.com |
| 16:46.13 | PatrickDK | I had horrible experiance with ICH and voice quiality |
| 16:46.17 | Zeeek | I believe they do DID all over the US (Voicepulse is limited) |
| 16:46.19 | *** join/#asterisk Moc (~mochouina@64.235.210.66) |
| 16:46.19 | *** join/#asterisk zeckill (~zec@221.124.100.43) |
| 16:46.20 | fa | bjohnson you know how to make a select to postgresel databxe form extesions.cond.. in mysql i did exten => s,3,MYSQL(Query resultid ${connid} SELECT\ `imie`\ FROM\ `user`\ WHERE$ |
| 16:46.25 | fa | ... |
| 16:46.27 | Bicster | iconecthere sucks ass |
| 16:46.33 | Moc | Bicster, they do |
| 16:46.38 | Zeeek | I'm sorry to hear that Patrick - I've never had a bad call with them |
| 16:46.40 | Bicster | I was one of their first abusers |
| 16:46.43 | vaewyn | iconnectnowhere is more like it |
| 16:46.56 | Moc | IgetScrewedNow |
| 16:46.57 | bjohnson | fa: I know how to use postgresql .. but I don't use it with * |
| 16:47.15 | Zeeek | haha - I don't know why it worked fine here - but I haven't used it for months |
| 16:47.25 | ManxPower | fa, Search the mailing list archives. Ask on the mailing list. |
| 16:47.27 | fa | bjohnson i heard that in asterisk is module.. in sandard bu t is comment, do you know sth about it? |
| 16:48.11 | Bicster | can voicepulse port a landline number? |
| 16:48.17 | `Sauron | Anyone here have the WiSIP phone? |
| 16:48.24 | Zeeek | Bicster ask them |
| 16:48.31 | ManxPower | Bicster, I don't trust any VoIP company to handle 911 |
| 16:48.49 | vaewyn | `Sauron: if you get it load the Zyxel firmware on it... the wisip firmware sucks big time |
| 16:48.55 | Bicster | well maybe I will just cut my analog line back to measured service then |
| 16:48.59 | bjohnson | `Sauron: that symbol in front is a pita |
| 16:48.59 | `Sauron | Hum. |
| 16:49.01 | `Sauron | Sucks |
| 16:49.04 | ManxPower | <PROTECTED> |
| 16:49.11 | ManxPower | That way I can keep my DSL too |
| 16:49.14 | `Sauron | They had it on sale for $199 this weekend |
| 16:49.18 | Bicster | my dsl is on a different pair (dry pair) |
| 16:49.20 | `Sauron | Now they're back to 249 |
| 16:49.26 | freat | is it just me, or does ulaw cause lots of problems? |
| 16:49.29 | `Sauron | I think the zyxel is more expensive |
| 16:49.32 | wasim | we're up to 365! and rising ... |
| 16:49.39 | ManxPower | freat, It's just you. Ulaw works fine. |
| 16:49.40 | tzanger | freat: it's you |
| 16:49.41 | Zeeek | degrees? |
| 16:49.42 | vaewyn | freat: if you don't have bandwidth it creates many problems :P |
| 16:49.50 | bjohnson | `Sauron: get a wifi pda and a bluetooth headset |
| 16:49.52 | freat | got plenty of bandwidth |
| 16:50.15 | freat | users report problems with audio though... and I've got it QoSd at highest prio |
| 16:50.22 | `Sauron | bjohnson: I already have a bluetooth headset |
| 16:50.22 | wasim | Zeeek: euros |
| 16:50.25 | freat | when I switch to GSM they are happy |
| 16:50.31 | `Sauron | But it doesn't let you enter numbers to dial with. :p |
| 16:50.46 | bjohnson | `Sauron: consider getting a pda with wifi and bt |
| 16:50.52 | vaewyn | Those new Hitachi (I think it was Hitachi) Wifi VoIP phones are supposedly good... I need to get my hands on one |
| 16:50.54 | `Sauron | Sigh. |
| 16:51.09 | doughecka_ | what the heck is yate? an asterisk wannabe? |
| 16:51.14 | `Sauron | the p910 already is a pda |
| 16:51.16 | bjohnson | `Sauron: they have sip softphones (and can keep appintments too!) |
| 16:51.17 | ManxPower | doughecka_, yes. |
| 16:51.17 | `Sauron | and it has bluetooth |
| 16:51.21 | fa | anyone have a example file to insert to pgsql for iax clients? |
| 16:51.28 | mutilator | for h323 tho isn't it |
| 16:51.28 | fa | i think about structure of database |
| 16:51.34 | Bicster | thanks ManxPower, Zeeek, bjohnson |
| 16:51.39 | `Sauron | You know, when someone asks for A, don't tell them to get B. |
| 16:51.41 | vaewyn | iPaq PDAs work great with iaxcomm :} |
| 16:51.50 | doughecka_ | ManxPower: interesting, has it reached a level where its 'good' to use like asterisk? |
| 16:51.54 | *** part/#asterisk Bicster (~Bicster@bicster.user) |
| 16:52.28 | `Sauron | And, *'s bluetooth support blows |
| 16:52.44 | ManxPower | doughecka_, I don't know. It's written by "lfy", who is a disgrunteld Asterisk user. lfy likes to come into the channel and bash Asterisk. |
| 16:52.52 | bjohnson | `Sauron: the bt would only be the last link .. not directly connected to * |
| 16:53.02 | doughecka_ | ah |
| 16:53.12 | ManxPower | doughecka_, lfy thinks Asterisk shoul dbe written in C++, so YATE is written in C++ |
| 16:53.21 | `Sauron | bjohnson: I already have a pda, and I already have a bt headset. |
| 16:53.24 | doughecka_ | whats asterisk written in? |
| 16:53.25 | ManxPower | Sorry, l-fy |
| 16:53.26 | doughecka_ | C? |
| 16:53.26 | bjohnson | `Sauron: here is answer to 'A' .. everybody I've heard talking about wifi phones wisjes they hadn't bought them |
| 16:53.38 | wasim | doughecka_: c |
| 16:53.39 | ManxPower | Asterisk is written in C, just like most things are. |
| 16:53.53 | doughecka_ | but see there, I didnt know, and I know asterisk... |
| 16:53.56 | eKo1 | C wasn't written in C. |
| 16:53.58 | doughecka_ | so it doesnt matter :) |
| 16:54.10 | bjohnson | eKo1: but some languages were |
| 16:54.13 | ManxPower | HMM? GCC is written in C! |
| 16:54.16 | `Sauron | bjohnson: That's probably because they couldn't afford what I can't afford either - the Cisco 7920's are great |
| 16:54.19 | `Sauron | but hella expensive |
| 16:54.38 | bjohnson | `Sauron: ManxPower is giving good reviews of the SPA 841s |
| 16:54.47 | eKo1 | C was written in B I think. |
| 16:54.55 | doughecka_ | ManxPower: are those phones good? |
| 16:54.58 | `Sauron | I don't think they're wireless? |
| 16:55.04 | doughecka_ | I am getting ready to quote a nice asterisk setup |
| 16:55.06 | bjohnson | `Sauron: btw .. I don't have a Cisco either |
| 16:55.07 | freat | ManxPower: dammit, our ISP and their stupid bandwidth manager |
| 16:55.08 | doughecka_ | and I need a good cheap phone |
| 16:55.29 | `Sauron | bjohnson: We have about 300 of them at work. They work great |
| 16:55.34 | ManxPower | I have not used the SPA-841 much, but what I have used it, it seems great. |
| 16:55.37 | bjohnson | `Sauron: correct .. no good reviews of ANY wireless voip phones |
| 16:55.39 | `Sauron | but the base price of $4-500/each |
| 16:55.40 | `Sauron | is a lot |
| 16:55.43 | ManxPower | I even upgraded to the 4-line version. |
| 16:55.54 | ManxPower | It's not a perfect phone, but it IS the BEST phone for under US$100 |
| 16:56.05 | doughecka_ | ManxPower: does it support all them cool multiple lines and stuff like that? |
| 16:56.08 | freat | wow that's a good price |
| 16:56.10 | doughecka_ | well* |
| 16:56.25 | ManxPower | doughecka_, 2-lines or 4-lines. Upgradable from 2 to 4 lines with a key code. |
| 16:56.31 | *** part/#asterisk humblast (~serty@212.247.174.226) |
| 16:56.34 | doughecka_ | free keycode? |
| 16:56.38 | ManxPower | 4-line upgrade is like $30 |
| 16:56.41 | doughecka_ | oh |
| 16:56.42 | djMax | anybody know the magic incantation to remove an old version of pwlib on Debian? |
| 16:56.53 | doughecka_ | well, if we need 4 line then they could up grade |
| 16:57.00 | `Sauron | bjohnson: Just get a big enough house, and you'll see |
| 16:57.01 | bjohnson | djMax: by the powers of Greystroke |
| 16:57.02 | `Sauron | or a business |
| 16:57.12 | doughecka_ | djMax: rm -rf / =D |
| 16:57.17 | ManxPower | bjohnson, I have L1: Business Line, L2: Personal Line, L3: Lover #1 Line, L4: Lover #2 Line |
| 16:57.22 | djMax | (djMax googles Greystroke :) ) |
| 16:57.24 | freat | lol |
| 16:57.27 | doughecka_ | ManxPower: HAH |
| 16:57.30 | bjohnson | `Sauron: ask ManxPower about wifi phones in business |
| 16:57.46 | `Sauron | Nevermind |
| 16:57.51 | Zeeek | Manx how do the 4 (or even 2) presences appear to asterisk? Like miltiple friend accounts? |
| 16:57.52 | doughecka_ | ManxPower: how does the multiple line thing work? |
| 16:57.53 | jaiger | djMax: dpkg --purge pwlib |
| 16:57.54 | ManxPower | bjohnson, If you continue to do that I shall be forced to put you on /ignore. |
| 16:57.58 | `Sauron | If you'd listen... |
| 16:58.05 | vaewyn | WiFi phones work... but you are gonna be eating battery all the time |
| 16:58.11 | ManxPower | doughecka_, Uh. The line rings. You pick it up. What do you want to know. |
| 16:58.19 | `Sauron | bjohnson: I already know all I need to know about voip/voipwifi phones in a business environment |
| 16:58.37 | doughecka_ | well, do they show up as seperate lines, like all 4 phones sees all 4 lines.. |
| 16:58.41 | `Sauron | We have over 300 phones, both wired and wifi |
| 16:58.41 | Zeeek | But Manx - does it have like 4 buttons? |
| 16:58.43 | ManxPower | doughecka_, yes. |
| 16:58.48 | doughecka_ | and each phone as one of each |
| 16:58.48 | ManxPower | zeckill, yes |
| 16:58.53 | freat | mmmm, buttons |
| 16:58.53 | `Sauron | I'm looking for something I can afford at home. |
| 16:58.55 | `Sauron | Sigh. |
| 16:59.02 | doughecka_ | and then you could see the status of the other phones |
| 16:59.05 | doughecka_ | just at a glance? |
| 16:59.13 | Zeeek | I'll get one in May based on your recco Manx |
| 16:59.15 | doughecka_ | or is there no status indicator? |
| 16:59.20 | ManxPower | I have no interest in seeing the status of other phones. |
| 16:59.24 | doughecka_ | I do |
| 16:59.40 | `Sauron | I'd just want the status of other lines on my phones |
| 16:59.42 | vaewyn | SIP doesn't have the base framework to do Line-in-use |
| 16:59.48 | `Sauron | that may or may not be shared across phones |
| 16:59.48 | doughecka_ | ah |
| 16:59.59 | psywar | hey how do I monitor whats going on in *? |
| 17:00.00 | Zeeek | so the main point is like X-Lite: you are talking, you can juggle several "lines" |
| 17:00.13 | ManxPower | vaewyn, The Hint() priority may or may not work for that. |
| 17:00.13 | psywar | I want to know what the CID is of people who call me, is there a log file or something? |
| 17:00.22 | jaiger | psywar: watch the logs or console |
| 17:00.25 | ManxPower | If I REALLY want to know the status of lines I use Flash Operator Panel. |
| 17:00.26 | freat | CID gets passed tothe phone |
| 17:00.27 | wasim | psywar: /var/log/asterisk/cdr-csv/blah |
| 17:00.32 | bjohnson | doughecka_: I read in the wiki that some SNOMs could map line in use to some of their lighted memory buttons |
| 17:00.32 | psywar | ty |
| 17:00.45 | doughecka_ | interesting |
| 17:00.47 | ManxPower | bjohnson, That's not a standard feature of phones. |
| 17:01.05 | ManxPower | the SPA-841 phones are obviously designed for the HOME user. |
| 17:01.07 | doughecka_ | I want it to work stably too, so if it doesnt quite work, then I will skip it |
| 17:01.30 | ManxPower | No power over Ethernet, no built in Switch Port, smallish (but OK) LCD display. |
| 17:02.00 | freat | doughecks_ : getting a stable phone is gonna be tough. most of us don't mind rebooting all phones when we come into work in the morning |
| 17:02.09 | ManxPower | But what do you expect for $80 |
| 17:02.35 | doughecka_ | I dont, since this is a client :) |
| 17:02.36 | fa | ManxPower I have ast_Data with pgsql.. but i need some examle of structur od table or isnert of some user? for iax2 |
| 17:02.41 | `Sauron | ManxPower: And getting into cisco's 7940/60 series has a whole different set of woes |
| 17:03.00 | ManxPower | fa, Let me say this again. I DON'T USE POSTGRESS. I DON'T USE DATABASES. NOW LEAVE ME ALONE. |
| 17:03.11 | doughecka_ | LOL |
| 17:03.11 | `Sauron | I tried to hack their call manager tftp boot stuff together to run them over skinny.. Interesting, to say the least. |
| 17:03.23 | jaiger | freat: what phone do you have that you reboot daily? |
| 17:03.35 | ManxPower | fa: If I have to I'm sure I can translate it into your native language. |
| 17:03.37 | Zeeek | GS |
| 17:04.01 | ManxPower | `Sauron, No sane person runs SCCP/Skinny with Asterisk. |
| 17:04.08 | `Sauron | I figured that out. |
| 17:04.11 | freat | jaiger: hahaa I was kidding |
| 17:04.19 | jaiger | I don't reboot my phones unless I'm messing with configs |
| 17:04.21 | jaiger | freat: ahh |
| 17:04.30 | freat | jaiger: we've got Polycom IP 500s they've been great |
| 17:04.34 | `Sauron | I was trying to throw together a config that would work w/o having to re-image the test 7940 that telecom borrowed me |
| 17:04.44 | jaiger | freat: that's what I have, we're happy so far |
| 17:04.47 | *** join/#asterisk multrix (~chatzilla@ALyon-252-1-23-55.w82-122.abo.wanadoo.fr) |
| 17:05.00 | `Sauron | We wanted to show the boss what OSS could do, instead of their $150k vendor solution. |
| 17:05.29 | vaewyn | speaking of GS... they pulled their last firmware from BETATEST... the Release notes are there but the firmware dissappeared. |
| 17:05.53 | `Sauron | Theo's SVN repository for chan_sccs is broken |
| 17:05.55 | fa | `Sauron can you give me some example of user iax in database - pgsql |
| 17:05.56 | fa | ? |
| 17:05.59 | `Sauron | chan_skinny doesn't work |
| 17:06.49 | `Sauron | fa: _IF_ I spent enough energy on it, yes. |
| 17:07.03 | fa | `Sauron fine, now? |
| 17:07.20 | *** join/#asterisk sjaak538 (~sjaaknabu@d5c53145.dsl.concepts.nl) |
| 17:07.28 | *** join/#asterisk gabb0 (~gabe@indo1.indosoft.unb.ca) |
| 17:08.35 | `Sauron | You're not a native english speaker, are you? |
| 17:08.50 | `Sauron | Ah, nope. |
| 17:09.29 | doughecka_ | hyperthreading with asterisk is ok? |
| 17:10.21 | *** part/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com) |
| 17:10.23 | stevekstevek | doughecka_: it used to cause problems with zaptel; don't know if that's presently the case.. |
| 17:11.22 | doughecka_ | ah, ok |
| 17:12.21 | *** join/#asterisk dca[laptop] (~trillian@sta-207-174-139-178.rockynet.com) |
| 17:12.29 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 17:13.28 | WildPikachu[BAR] | hrmmmm |
| 17:13.55 | WildPikachu[BAR] | i have a capi device, set everything up... but when i dail in it doesn't seem anything detects the ring |
| 17:14.07 | *** join/#asterisk tsimshatsui (~BBRodrigu@pD9EA7084.dip.t-dialin.net) |
| 17:14.11 | Manipura | http://www.ebaumsworld.com/bananaphone.html |
| 17:14.43 | Sedorox | <PROTECTED> |
| 17:14.43 | Sedorox | Jan 31 10:11:46 WARNING[91105]: chan_iax2.c:7194 find_cache: Timeout waiting for ss-server:xxxxxx@stormy.smart-serv.net/local exten 1000 |
| 17:14.43 | Sedorox | <PROTECTED> |
| 17:15.04 | tsimshatsui | hi people, i hv germany mobile route - 0.07Euro / Any quantity / 8 sec. PDD / No CLI, ready to test now, anyone interested ? |
| 17:15.31 | PakiPenguin | uk == 240v? or EU == 240v ? |
| 17:15.57 | *** join/#asterisk mAsH` (~mAsH@host46-29.pool8173.interbusiness.it) |
| 17:17.08 | *** join/#asterisk denon (denon@synapse.subneural.net) |
| 17:17.08 | *** mode/#asterisk [+o denon] by ChanServ |
| 17:17.09 | Delvar | PakiPenguin: yes, uk/eu are compatable, you can even jerry rig an EU psu into a UK wall socket if you use a skrewdriver in the top (ground) hole |
| 17:17.38 | Delvar | PakiPenguin: of cource i dont suggest you do it like that.. get an adapter :) |
| 17:18.08 | *** join/#asterisk AsteriskNooB (AsteriskNo@207-114-232-10.gen.twtelecom.net) |
| 17:18.16 | AsteriskNooB | morning everyone |
| 17:18.35 | WildPikachu[BAR] | hrmmmm |
| 17:18.48 | Sedorox | whats error 484 mean? |
| 17:19.42 | *** part/#asterisk zbysio (~zbysio@chello084010031149.chello.pl) |
| 17:21.15 | *** join/#asterisk harryvv (~noyb@S010600055d210201.vs.shawcable.net) |
| 17:25.57 | PakiPenguin | is voip-info.org down? |
| 17:26.01 | PakiPenguin | its horribly slow for me |
| 17:26.09 | Sedorox | I can't get onto it |
| 17:26.48 | *** join/#asterisk sarumont (~sarumont@pcp04563795pcs.dallas01.ga.comcast.net) |
| 17:26.56 | *** part/#asterisk sarumont (~sarumont@pcp04563795pcs.dallas01.ga.comcast.net) |
| 17:27.31 | Zeeek | u are talking, you can juggle several "lines" |
| 17:27.31 | Zeeek | GS |
| 17:27.36 | Zeeek | whoops |
| 17:27.57 | Zeeek | What was that site that always hed the latest GS firmware? siphello.com ? |
| 17:28.04 | Zeeek | hello.de? |
| 17:28.05 | *** join/#asterisk jpmcallister (~jpmcallis@kapla.escelsa.com.br) |
| 17:28.11 | jpmcallister | <PROTECTED> |
| 17:28.23 | WildPikachu[BAR] | :( |
| 17:28.28 | Zeeek | nah that wasn't uit |
| 17:28.31 | WildPikachu[BAR] | seems my mISDN device not working |
| 17:28.55 | Schism | Sedorox: what protocol threw that error? |
| 17:29.21 | Sedorox | iax(2) |
| 17:29.23 | PakiPenguin | umm |
| 17:29.37 | PakiPenguin | voip-info.org has it |
| 17:29.58 | Zeeek | what the site? |
| 17:31.03 | PakiPenguin | grandstream , check for it |
| 17:31.27 | *** join/#asterisk file (~symlink@mctn1-142166196242.nb.aliant.net) |
| 17:31.27 | Schism | it appears to be address incomplete |
| 17:31.30 | Zeeek | yes I'm looking at 100 pages - just curious and thought someone had it on the tipof their tongue |
| 17:31.41 | Schism | <PROTECTED> |
| 17:31.45 | Sedorox | hmmm |
| 17:31.47 | Schism | http://www.voip-info.org/wiki-SIP+response+class4 |
| 17:31.54 | PakiPenguin | just a sec |
| 17:32.04 | *** join/#asterisk crash3m (crash3m@crash3m.user) |
| 17:32.09 | harryvv | never seen this before modoprobe wcfxo modprobe zaptel and it spews out lots of unresolved symbols. |
| 17:32.19 | Sedorox | I kinda got it working now.... 'cept that when I dialed it.. I got like a "machine gun sound" which when I used X-Lite and got that.. it was codecs.. but I have allow=all..... |
| 17:32.21 | *** join/#asterisk JunK-Y (~grepmoo@65.39.228.5) |
| 17:32.27 | Schism | harryvv: u probably need to depmod -ae it |
| 17:32.31 | Zeeek | hellofone.com |
| 17:32.35 | Schism | harryvv: or recompile them |
| 17:32.47 | PakiPenguin | http://www.hellofone.com/downloads.html |
| 17:32.55 | AsteriskNooB | anybody know hwo to put a pause in a dialstring? |
| 17:32.55 | harryvv | yea it was compiled and working yesterday |
| 17:32.59 | harryvv | thats whats strange |
| 17:33.00 | Schism | Sedorox: u got verbosity on 3 on the console? |
| 17:33.08 | Sedorox | yea.. on both console's |
| 17:33.16 | Schism | wierd |
| 17:33.19 | Sedorox | I start with -vvvc in a screen session |
| 17:33.46 | Schism | and it doesn't spew anything on the console about codec mismatch? |
| 17:34.03 | Sedorox | -- Call accepted by 24.71.218.177 (format unknown) |
| 17:34.03 | Sedorox | then |
| 17:34.09 | Sedorox | -- Call accepted by 24.71.218.177 (format g723) |
| 17:34.16 | Sedorox | <PROTECTED> |
| 17:34.31 | Sedorox | and on the other side.. the only thing I see is |
| 17:34.31 | Schism | interesting |
| 17:34.33 | Sedorox | -- Accepting AUTHENTICATED call from 64.251.71.178, requested format = 1, actual format = 1 |
| 17:34.38 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 17:34.44 | Schism | hmm |
| 17:34.49 | WildPikachu[BAR] | anyone here use capi? |
| 17:35.01 | wasim | WildPikachu[BAR]: kape does capi |
| 17:35.05 | Sedorox | ok.. disallow=all with allow-gsm works |
| 17:35.11 | Schism | :) |
| 17:35.50 | Sedorox | which has a lower speed? gsm or g723? |
| 17:36.03 | Sedorox | bandwidth... |
| 17:36.08 | Schism | g723 |
| 17:36.12 | Sedorox | thought so... |
| 17:36.16 | Schism | gsm is 90k w/ overhead |
| 17:36.27 | Schism | g723 is like 33k? w/ overhead |
| 17:36.41 | Sedorox | wow... |
| 17:36.47 | Sedorox | big diff |
| 17:36.52 | Sedorox | yea.. kinda wanna save on BW.. so... |
| 17:37.04 | Delvar | thought gsm was a lot lower than that? |
| 17:37.17 | Schism | the protocol is 64k |
| 17:37.33 | WildPikachu[BAR] | kape? |
| 17:37.35 | WildPikachu[BAR] | wasim, kape? |
| 17:37.38 | Schism | but the setup and stuff adds to it |
| 17:37.47 | Sedorox | when I have allow=g732.1 |
| 17:37.49 | Schism | sorry, the codec is 64k |
| 17:37.49 | Sedorox | -- Format for call is g723 |
| 17:37.53 | *** join/#asterisk ta[i]nted (~ta_i_nted@65-60-70-242-cust.telepacific.net) |
| 17:37.54 | Sedorox | I get the machine gun affect.... |
| 17:38.06 | harryvv | schism, yea take a look at this. :) http://pastebin.ca/5038 |
| 17:38.08 | *** join/#asterisk dtwilson (~dave@host217-36-121-129.in-addr.btopenworld.com) |
| 17:38.13 | Sedorox | whats another good low-speed codec? |
| 17:38.16 | Delvar | i know that :).. i just thought it was more 40k...nm im probably miss informed |
| 17:38.20 | PakiPenguin | Zeeek: which firmware are you on? |
| 17:38.26 | *** join/#asterisk jsmith (~jsmith@pip.drs1.omniture.com) |
| 17:38.36 | Delvar | g729 is prety good :) |
| 17:38.37 | Zeeek | 5.11 - I can't go any further |
| 17:38.40 | *** part/#asterisk jsmith (~jsmith@pip.drs1.omniture.com) |
| 17:38.48 | Zeeek | I have them all up to 20 or maybe even 21 |
| 17:38.49 | Sedorox | how is it on BW? like what's it speed? |
| 17:38.54 | Schism | harryvv: have you looked in your dmesg? |
| 17:39.03 | harryvv | no have not |
| 17:39.07 | mAsH` | i'm starting * and i get this errore |
| 17:39.09 | Zeeek | No I have 5.22 - I haven't tried ti yet |
| 17:39.15 | mAsH` | anyoneone can help me? |
| 17:39.24 | mAsH` | <PROTECTED> |
| 17:39.24 | mAsH` | <PROTECTED> |
| 17:39.24 | mAsH` | <PROTECTED> |
| 17:39.24 | mAsH` | <PROTECTED> |
| 17:39.24 | mAsH` | Illegal instruction (core dumped) |
| 17:39.38 | dtwilson | hi all - would anybody know why I wouldnt seem to have iax2 nor iax commands available from CLI? its an inherited box I'm looking at |
| 17:39.52 | Delvar | http://www.voip-info.org/wiki-Bandwidth+consumption |
| 17:40.15 | Delvar | G.729 8 Kbps 31.2 Kbps |
| 17:40.15 | Zeeek | dtwilson not built and/or not loaded |
| 17:40.31 | `Sauron | Dum di dum. |
| 17:40.38 | dtwilson | Zeeek: I was thinking that - but surely iax.conf wouldnt exist if that was the case? |
| 17:40.49 | Sedorox | <PROTECTED> |
| 17:40.51 | Zeeek | sho wmodules |
| 17:40.53 | Sedorox | doesn't seem too bad |
| 17:40.53 | JunK-Y | dtwilson: asterisk*CLI> show modules like iax2 |
| 17:40.54 | *** join/#asterisk orpheusnickinuse (~orpheus@router.emperor-sw2.exsbs.net) |
| 17:41.04 | Zeeek | the conf doesn't mean anything |
| 17:41.39 | WildPikachu[BAR] | this is weird |
| 17:41.45 | WildPikachu[BAR] | i turned on all debugging i could find |
| 17:41.50 | Schism | sorry, I was gsm = g.711 |
| 17:42.00 | WildPikachu[BAR] | when i dial the number, it just rings... nothing happening |
| 17:42.14 | Zeeek | dtwilson show modules iax2 and sip will be near the top |
| 17:42.20 | orpheusnickinuse | anyone had any experience with broadvoice? (i know the wiki says how to get thier service to work, i mean experience with the company in general) |
| 17:42.20 | dtwilson | ahha - iax modules not there - now - how to add it in? |
| 17:42.30 | harryvv | schism, i just looked though my dmesg. what are you asking. |
| 17:42.32 | PakiPenguin | Can i get some volume discount on g729 |
| 17:42.42 | Schism | harryvv: did it do a core dump? |
| 17:42.42 | Zeeek | look in modules.conf and see if it's not loaded on purp |
| 17:42.52 | dtwilson | Zeek: cheers |
| 17:42.54 | Schism | harryvv: did you see anything regarding the modules? |
| 17:43.00 | harryvv | no |
| 17:43.02 | Zeeek | with a noload= |
| 17:43.15 | Zeeek | maybe grep noload modules.conf even |
| 17:43.31 | multrix | I got free access on a SIP proxy where I can call everywhere in europe canada and US, do you think its illegal or dangerous ? |
| 17:43.31 | Schism | harryvv: I would rmmod the modules, and then recompile them |
| 17:43.36 | Delvar | http://www.terracall.com/FAQs_white_1.aspx << prety good table |
| 17:43.36 | harryvv | okay |
| 17:43.44 | harryvv | make; make install |
| 17:43.45 | JunK-Y | dtwilson: load chan_iax2.so |
| 17:43.47 | harryvv | okay |
| 17:44.20 | harryvv | modules are not loaded according to rmmod |
| 17:44.35 | dtwilson | JunK-Y: I just commented out the noload = chan_iax2.so - so I need an actual "load = chan_iax2.so" line then? |
| 17:44.45 | Schism | just wanted to make sure |
| 17:44.48 | Zeeek | dtwilson no retart and see |
| 17:44.54 | PakiPenguin | mohaa time , brb |
| 17:44.59 | dtwilson | Zeek I did reload |
| 17:45.06 | dtwilson | still no iax2 command |
| 17:45.30 | Delvar | when changing modules dont you have to stop/start asterisk? |
| 17:45.32 | JunK-Y | dtwilson: try to restart now. |
| 17:45.45 | JunK-Y | ya dont have to have a load chan_iax2 line in ur module |
| 17:45.52 | JunK-Y | until ya've the .so created. |
| 17:46.03 | harryvv | Schism, here is what I got. http://pastebin.ca/5039 |
| 17:46.03 | JunK-Y | in /usr/lib/asterisk/modules/chan_iax2.so |
| 17:46.20 | dtwilson | waiting for restart to finish |
| 17:46.42 | Schism | hmm |
| 17:47.08 | *** join/#asterisk miguellinux (~miguel@mail.cajonperuano.com) |
| 17:47.12 | Schism | rmmod the modules w/ unresolved symbols and then do a "make clean; make install" |
| 17:47.23 | Schism | rsorry |
| 17:47.25 | Schism | rm the modules |
| 17:47.28 | Schism | not rmmod :-P |
| 17:47.32 | harryvv | okay |
| 17:47.47 | harryvv | like remove this one |
| 17:47.53 | JunK-Y | asterisk modules != kernel modules. |
| 17:48.08 | Sedorox | yay... two asterisk's connected and working... |
| 17:48.09 | Schism | zaptel modules = kernel modules |
| 17:48.20 | harryvv | ie, remove the modules in those last lines of the url i just showed you right? |
| 17:49.05 | Zeeek | dtwilson - we're hanging on a thread here! |
| 17:49.25 | Schism | harryvv: yes |
| 17:49.29 | harryvv | okay |
| 17:49.33 | dtwilson | restart seems to be taking forever |
| 17:49.38 | Schism | that have unresolved symbols |
| 17:49.44 | Zeeek | dtwilson not a great sign |
| 17:49.46 | *** join/#asterisk jterrero (~some@66.28.34.162) |
| 17:49.58 | dtwilson | Zeeek: I know :( - am worried now |
| 17:50.04 | Zeeek | maybe the iax.conf is screwed up tho |
| 17:50.10 | jterrero | can someone help me out? |
| 17:50.11 | jterrero | Jan 31 12:46:53 WARNING[17722]: file.c:475 ast_openstream: File voicemail//300/greet does not exist in any format |
| 17:50.11 | jterrero | Jan 31 12:46:53 WARNING[17722]: file.c:779 ast_streamfile: Unable to open voicemail//300/greet (format ilbc): No such file or directory |
| 17:50.16 | Zeeek | make sure it has the right port and all that |
| 17:50.29 | dtwilson | 5036? |
| 17:50.35 | *** join/#asterisk nitram (nitram@superblob.com) |
| 17:50.36 | Zeeek | 4569 IIRC |
| 17:50.43 | benjr | Is there a way to use encrypted passwords in SIPFRIENDS MYSQL table? anyone, please |
| 17:50.43 | Zeeek | 5036 was IAX |
| 17:51.06 | dtwilson | don't think iax.conf is the problem tho - asterisk just doesnt seem to start up again :S |
| 17:51.16 | Sedorox | bbl |
| 17:51.17 | Zeeek | where diod it stop? |
| 17:51.18 | Sedorox | class |
| 17:51.33 | dtwilson | soon as I did restart now |
| 17:51.44 | Zeeek | oh. |
| 17:51.59 | Zeeek | drums stop. Not good. |
| 17:52.23 | Zeeek | as in King Kong |
| 17:52.24 | Delvar | benjr: md5secret ? |
| 17:52.30 | Zeeek | "Drums Stop. No Good" |
| 17:52.35 | *** join/#asterisk RoyK (~roy@i171219.dsl.tjukkband.no) |
| 17:52.39 | harryvv | schism, thay seem to be comming back after i rm them |
| 17:52.50 | Schism | yes, after you do a make install |
| 17:52.52 | Sedorox | thanks for your help Schism |
| 17:52.53 | Schism | right? |
| 17:52.53 | RoyK | rm -f harryvv |
| 17:53.00 | Schism | Sedorox: n/p |
| 17:53.07 | Sedorox | ttyl.... off to a boring class |
| 17:53.11 | dtwilson | hmmm weird - just returned modules.conf to original and its back up again |
| 17:53.11 | harryvv | k |
| 17:53.22 | Schism | you are removing the evil modules |
| 17:53.27 | Schism | and recompiling them :) |
| 17:53.29 | dtwilson | so, mustnt have chan_iax2.so installed |
| 17:53.30 | Zeeek | dtwilson sounds like you need some remaking to be done |
| 17:53.31 | Schism | do a depmod -a |
| 17:53.39 | Schism | after you do a make install |
| 17:53.43 | Schism | then try to modprobe them |
| 17:54.38 | dtwilson | Zeeek: I'm worried about doing a remake, cos its a production box which has soem extensive python customization stuff in it |
| 17:55.09 | Zeeek | Be afraid! Be very afraid! |
| 17:55.24 | Zeeek | When the prod box goes down, so do you! |
| 17:55.30 | harryvv | well, I rm -f all those lines by simply copping them and then pasting after rm -f and thay still come back after /usr/src/zaptel make; make install. so no chance of typos introduced. |
| 17:55.31 | dtwilson | indeed |
| 17:55.36 | Zeeek | Plus it made me feel like shit for 24 hours |
| 17:55.51 | harryvv | okay |
| 17:55.52 | harryvv | depmode |
| 17:55.54 | WildPikachu[BAR] | :(((((( |
| 17:55.56 | Schism | harryvv: that is good |
| 17:56.01 | Schism | harryvv: u want them to come back |
| 17:56.03 | Zeeek | knowing my SIP and IAX stuff at home was DEAD |
| 17:56.12 | *** join/#asterisk BoRiS (~boris@S01060040ca1e5b54.wp.shawcable.net) |
| 17:56.56 | PakiPenguin | Zeeek: know anything about iax and firewall ( what ports i need forwarded to my machine to make it work ) |
| 17:57.05 | *** join/#asterisk ard (~ard@2001:7b8:32d:0:20c:6eff:fe18:d11f) |
| 17:57.10 | BoRiS | hi guys |
| 17:57.10 | multrix | can somebody tell me about this ? : I got a free access to a SIP proxy, and I can call all europe, canada and US for free.... what could it be behind this ?? hacked pbx ? |
| 17:57.11 | Zeeek | you shouldn't need to forward for IAX |
| 17:57.41 | Zeeek | what IAX device are you using PakiP? |
| 17:57.48 | Delvar | multrix: bad config on proxi |
| 17:57.49 | file[laptop] | BoRiS!!!!!!!!!!!!!!!!!!!!! |
| 17:57.56 | BoRiS | file!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!! Wassssssssssssup? |
| 17:58.01 | file[laptop] | de nada, u? |
| 17:58.07 | harryvv | okay schisim im getting the top three lines as symbol errors when doing depmod -a |
| 17:58.19 | Schism | how about depmod -ae |
| 17:58.19 | jterrero | can someone please help me out? i have my DID being forwarded to extension 300, in voicemail.conf i have "300 => 2121, Some D. Dude," when i call the number my phone rings and after time out i expect it to go to voicemail, in asterisk debug i get the following message.. does anyone know what my prob might be ? |
| 17:58.20 | jterrero | Jan 31 12:46:53 WARNING[17722]: file.c:475 ast_openstream: File voicemail//300/greet does not exist in any format |
| 17:58.20 | jterrero | Jan 31 12:46:53 WARNING[17722]: file.c:779 ast_streamfile: Unable to open voicemail//300/greet (format ilbc): No such file or directory |
| 17:58.26 | PakiPenguin | Zeeek: iaxphone ( everything's blocked , i need to specify ports to have them open, work firewall ) |
| 17:58.35 | harryvv | okay try that |
| 17:58.43 | BoRiS | Not too much, working on some brochures and pricing stuff :) |
| 17:58.47 | Zeeek | 4569 PakiP |
| 17:58.48 | file[laptop] | BoRiS: ah |
| 17:58.50 | Schism | harryvv: I dunno, have you tried to reboot your machine? |
| 17:59.02 | harryvv | scisim, depmod -ae worked |
| 17:59.03 | *** join/#asterisk FuzzyCat (~ScaredyCa@84.119.133.131) |
| 17:59.08 | harryvv | no erorrs |
| 17:59.14 | Schism | sweet! |
| 17:59.14 | Schism | :) |
| 17:59.21 | Schism | try to modprobe them |
| 17:59.24 | file[laptop] | BoRiS: Space changed their times and stuff... |
| 17:59.24 | harryvv | yea, no fricken idea what caused this |
| 17:59.25 | PakiPenguin | Zeeek: udp /tcp? what about others 10000 and ? |
| 17:59.30 | Zeeek | jterro the double slash may not be a good omen - other than that maybe no file exists? |
| 17:59.39 | Zeeek | no just 4569 UDP AFAIK |
| 17:59.49 | Zeeek | 10000 is for SIP |
| 18:00.02 | Zeeek | I theeeeeeenk |
| 18:00.12 | harryvv | schism, get 13 unresolved symbols when doing modprobe wcfxo |
| 18:00.13 | jterrero | Zeeek: what file does not exist? ive only been working with iax.conf, extensions.conf and voicemail.conf |
| 18:00.17 | SuPrSluG | any dundi people here? |
| 18:00.24 | Schism | harryvv: even after depmod -ae worked? :-( |
| 18:00.26 | Zeeek | the file it's compklaining about |
| 18:00.30 | harryvv | yea |
| 18:00.31 | dtwilson | Zeeek: yayy - it was my bad iax.conf |
| 18:00.40 | Schism | harryvv: have you tried to reboot your server? |
| 18:00.43 | harryvv | no |
| 18:00.46 | Zeeek | jterrero - it needs a greetng file for voicemail and it isn't finding it |
| 18:00.46 | harryvv | let me do that |
| 18:00.52 | *** join/#asterisk ard (~ard@2001:7b8:32d:0:20c:6eff:fe18:d11f) |
| 18:01.00 | jterrero | Zeeek: doesnt asterisk have a default file? |
| 18:01.00 | Zeeek | dtwilson so even a stopped clock (me) can be right twice a day |
| 18:01.11 | harryvv | rebooting |
| 18:01.13 | Zeeek | yes but it isn't finding it : THINK now |
| 18:01.57 | *** join/#asterisk ard (~ard@2001:7b8:32d:0:20c:6eff:fe18:d11f) |
| 18:02.08 | dtwilson | Zeeek: if you turn the clock round, it can be right several times a day |
| 18:02.27 | Zeeek | Been know to happen :) |
| 18:02.59 | modulus_ | just got head |
| 18:03.01 | Zeeek | INBOX busy.gsm greet.gsm greet.wav |
| 18:03.03 | modulus_ | w00t! |
| 18:03.16 | modulus_ | Connected to Asterisk CVS-HEAD-01/31/05-09:56:13 currently running on asterisk (pid = 13001) |
| 18:03.47 | Zeeek | jterrero : here's what I have for example root@r:/etc/asterisk# ls /var/spool/asterisk/voicemail/default/2000 |
| 18:03.54 | Zeeek | INBOX busy.gsm greet.gsm greet.wav |
| 18:04.22 | file[laptop] | BoRiS: Which TNG is on? |
| 18:04.30 | *** part/#asterisk JunK-Y (~grepmoo@65.39.228.5) |
| 18:04.43 | PakiPenguin | i love TNG!!! |
| 18:04.49 | *** join/#asterisk clive- (~pirch@rrba-146-64-178.telkomadsl.co.za) |
| 18:04.57 | *** join/#asterisk machinehd (~machinehd@storm.bcgroup.net) |
| 18:04.58 | BoRiS | Something with a ferengi coming on board...looks like trouble to me |
| 18:05.09 | PakiPenguin | argh tng == the next generation |
| 18:05.13 | benjr | delvar: md5secret where? there's not a column with that name |
| 18:06.11 | BoRiS | I love TNG also |
| 18:06.18 | file[laptop] | erm |
| 18:06.47 | fa | `Sauron so? Do you have some good structure and data? |
| 18:06.54 | modulus_ | jbot g-g-g-g? |
| 18:06.55 | jbot | G-UNIT! |
| 18:07.00 | modulus_ | jbot g-unit? |
| 18:07.01 | jbot | g-unit stands for "Guerilla Unit". It's members are Tony Yayo, Lloyd Banks, Young Buck, and the leader 50 Cent. Their official DJ is DJ Whoo Kid. Also see http://www.g-unitsoldier.com/ |
| 18:07.06 | *** join/#asterisk Gerrath (~Gerrath@shanev.lifecor.com) |
| 18:08.07 | Zeeek | Who is the captain in TNG? |
| 18:08.16 | modulus_ | jean luc picard |
| 18:08.22 | modulus_ | dumb ass |
| 18:08.25 | file[laptop] | oh this episode, I don't like it |
| 18:08.42 | Zeeek | I like the later series with the girls with big lips |
| 18:08.47 | WildPikachu[BAR] | is there a way to make a call using capi manually? |
| 18:08.50 | WildPikachu[BAR] | to test if the line is working |
| 18:08.51 | netsurfer | I dont fully understand how the contexts work in extensions.conf - if a user has context=foo in sip.conf, do they have permissions for everything below [foo] or just [foo] |
| 18:08.52 | machinehd | I have 2 boxes with what I believe is the same config. One has * bound to "127.0.0.1:5038" the other is "10.10.200.5:5038"... sip.conf is identical. Any ideas why? |
| 18:09.11 | Zeeek | foo and what's included in foo, that's all |
| 18:09.18 | netsurfer | ok |
| 18:09.20 | modulus_ | machinehd, /etc/asterisk/manager.conf |
| 18:09.23 | Zeeek | which is the point |
| 18:09.29 | modulus_ | machinehd, specifies which port/ip to listen on |
| 18:09.35 | netsurfer | and if context=foo;foobar they have permissions for both of those ? |
| 18:09.41 | machinehd | modulus_, they are identical |
| 18:09.48 | *** join/#asterisk yogurt2ungue (~charlie@203-132-126-200.fibertel.com.ar) |
| 18:09.54 | Zeeek | you can't put multiple contexts that I know of |
| 18:10.07 | Zeeek | but I know nothing, nothing... |
| 18:10.37 | Zeeek | An excellent doc on contexts: http://asteriskdocs.org |
| 18:11.08 | *** join/#asterisk ard (~ard@2001:7b8:32d:0:20c:6eff:fe18:d11f) |
| 18:11.19 | netsurfer | yes.. been reading for days on it, just couldnt find an answer to that question anywhere.. the docs kind of assume u just know |
| 18:11.31 | netsurfer | or maybe im just a bit dumb :P |
| 18:11.32 | Zeeek | No it's described in detail there |
| 18:11.35 | Nugget | context= doesn't give permission, it sets the default context. there can only be one default. |
| 18:11.36 | modulus_ | LIGHTS OUT! |
| 18:11.53 | dtwilson | I cant see |
| 18:11.55 | netsurfer | Nugget - I dont mean in [general] |
| 18:12.01 | Nugget | neither do I. |
| 18:12.03 | netsurfer | Nugget - I mean in [user] |
| 18:12.06 | Nugget | so do I. |
| 18:12.10 | Zeeek | "Introduction to Creating Dialplans" |
| 18:12.24 | Zeeek | "This file is made up of four main parts: contexts , extensions , priorities , and applications ." etc |
| 18:12.32 | netsurfer | fa - ask before pm'ing,. im busy right now |
| 18:12.43 | Zeeek | Contexts play an organizational role within the dialplan. Contexts also define scope. " |
| 18:12.48 | Zeeek | etc |
| 18:13.02 | Zeeek | http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html_one/vm1.html |
| 18:13.15 | netsurfer | i'll have another read.. im sure its all in the docs, maybe I just dont understand it fully |
| 18:13.25 | Zeeek | It's right there |
| 18:13.41 | Zeeek | they developed that doc over many months - it's excellent |
| 18:13.58 | fa | are anyone using postgresql to IAX users? |
| 18:14.01 | fa | od sip? |
| 18:14.21 | Nugget | there are no access controls on contexts. all calls have the same access to all contexts. it's up to you to make sure that there's no path from a connection's default context to a context which it should be able to access. |
| 18:14.44 | Nugget | if a call's default context has no path to another context, then there's no way for the call to hop to that context |
| 18:14.50 | Zeeek | then how can it access what it'sz supposed to access? :) |
| 18:15.20 | Zeeek | "make sure that there's no path from a connection's default context to a context which it should be able to access" |
| 18:15.29 | Zeeek | no wonder he's confused :) |
| 18:15.32 | Nugget | heh |
| 18:15.57 | Zeeek | just make sure you do not have the keys to the car you wish to start |
| 18:16.17 | Zeeek | that *is* paranoid security ! |
| 18:16.20 | Nugget | a better analogy would be "make sure that no roads lead to the place you don't want the driver to go." :) |
| 18:16.26 | netsurfer | http://www.voip-info.org/wiki-Asterisk+RealTime+Sip <-- I was ok til I read this lol |
| 18:16.49 | netsurfer | it shows a user with 2x context= |
| 18:16.49 | Zeeek | or even the non poetic but literal "Don't include dangerous contexts in insecure ones" |
| 18:17.11 | Zeeek | You use RealTime ? |
| 18:17.21 | Nugget | that page is, generously speaking, quite misleading. |
| 18:17.40 | netsurfer | Zeeek - I am thinking of implementing it, but at the moment, no |
| 18:17.57 | netsurfer | however it shows a standard sip.conf entry |
| 18:18.12 | Zeeek | you mean Extconfig.conf Setup ? |
| 18:18.18 | Nugget | a sip.conf entry with two context lines is not standard, it's malformed. |
| 18:18.37 | netsurfer | then that page is very misleading |
| 18:18.40 | Nugget | it is. |
| 18:18.58 | Zeeek | Ya it doesn't apply to the normal setup without realtime |
| 18:19.08 | Zeeek | walk before you run, I guess |
| 18:19.21 | Zeeek | or run and bust yer ass like my grandson does |
| 18:19.42 | Zeeek | I wonder if there are kids that have never had chin stitches ? |
| 18:19.46 | netsurfer | well, in realtime then it shows the 2 contexts stored as default;local which I found odd |
| 18:20.02 | Zeeek | worry about it when you actually HAVE realtime |
| 18:20.06 | netsurfer | anyway.. thx for clearing that matter up :) |
| 18:20.07 | Nugget | netsurfer: not any more. :) |
| 18:20.12 | Zeeek | heh |
| 18:20.20 | netsurfer | Zeeek - I have it now.. but im not using it with sip.conf |
| 18:20.45 | Zeeek | you have one on me because I can't imagine why that would be of interest :) |
| 18:20.45 | *** join/#asterisk buddah (~hnic@208.179.86.5) |
| 18:21.11 | netsurfer | Nugget - thats clearer :) |
| 18:21.47 | netsurfer | Nugget - "in the column it should be separated by a semicolon. For example, an entity that looks like: " |
| 18:21.54 | Zeeek | why am I not seeing a difference? |
| 18:21.56 | netsurfer | now there's no semicolon ;) |
| 18:22.09 | netsurfer | Zeeek - refresh? |
| 18:22.27 | netsurfer | kick ur webcache :oP |
| 18:22.33 | Zeeek | where di dit change? |
| 18:22.43 | netsurfer | took one of the context= out |
| 18:22.52 | Zeeek | what is the heading above? |
| 18:23.00 | netsurfer | and removed ;local from the db entry |
| 18:23.35 | Zeeek | I must be on the wrong page |
| 18:24.09 | netsurfer | Zeeek http://www.voip-info.org/wiki-Asterisk+RealTime+Sip |
| 18:24.19 | Nugget | ok, updated. thanks netsurfer. |
| 18:24.23 | netsurfer | yw |
| 18:24.39 | Zeeek | that took about 10 refreshes |
| 18:24.53 | netsurfer | lol crappy isp ;) |
| 18:25.15 | Zeeek | I'm always tempted to tweek and use stuff like realtime but I think for like 3 users why run mysql etc |
| 18:25.29 | Nugget | I'd be more tempted if it were postgresql. |
| 18:25.31 | netsurfer | because its fun getting it to work :oP |
| 18:25.42 | *** join/#asterisk george_ (~george@216.157.203.105) |
| 18:25.43 | Zeeek | yes that's why I have * in the first place |
| 18:25.54 | george_ | Hello everyone! |
| 18:25.58 | Zeeek | but I'm painted in a corned now that it is used as our production pbx |
| 18:26.14 | george_ | I have a quesion about gentoo, *, and MOH.... |
| 18:26.14 | Zeeek | it's gorgeous George! |
| 18:26.27 | Zeeek | oh, maybe nbot |
| 18:26.29 | netsurfer | if u dont use mpg123 that is ;) |
| 18:26.41 | Nugget | everyone should add enum e164.org lookups to their dialplan today. |
| 18:26.48 | Nugget | it's good karma |
| 18:26.57 | george_ | Um, I am trying to use mpg123, is that not what I should do for MOH? |
| 18:26.58 | *** join/#asterisk Luhiwu (~marsosa@200.63.87.246) |
| 18:27.02 | Luhiwu | hi all |
| 18:27.05 | Zeeek | yes but I was getting no register from e164.org all day so I removed it |
| 18:27.10 | Nugget | pout |
| 18:27.14 | netsurfer | I keep getting stray mpg123 processes starting up without putting anyone on hold |
| 18:27.25 | netsurfer | earlier I did ps x and had like 12 of them |
| 18:27.30 | Zeeek | they're not stray |
| 18:27.38 | george_ | If I have an mp3 in the spec'd directory, mpg123 will start, but * will not. |
| 18:27.51 | netsurfer | I didnt have 12 ppl on hold.. in fact not even one ;) |
| 18:27.58 | Zeeek | netsurfer you should have about 6 |
| 18:28.05 | Zeeek | at startup |
| 18:28.05 | modulus_ | redhat sure is stupid |
| 18:28.08 | jterrero | anyone know where i can find some docs on flashing Cisco 79xx phones to use SIP ? |
| 18:28.09 | Luhiwu | anyone is using chan_h323 for incoming calls? |
| 18:28.13 | netsurfer | oh, I only have 2 right now |
| 18:28.17 | modulus_ | i switch to runlevel3 and it calls runlevel6 first |
| 18:28.21 | modulus_ | stuuuuuuuuuupid |
| 18:28.31 | Zeeek | under certain conx if you stop * you'll have a bunch of mp3 processes running |
| 18:28.33 | george_ | I have mpg123 v.059s, but that's what I have on my suse box. |
| 18:28.34 | Nugget | set the default runlevel to 6 in inittab. :) |
| 18:28.55 | george_ | nad the suse box works fine. |
| 18:28.59 | *** join/#asterisk gustavoz (~gustavoz@gustavoz.developer.gentoo) |
| 18:29.12 | *** join/#asterisk tclark (~TC@S0106000c413a1c61.gv.shawcable.net) |
| 18:29.38 | fa | anyone try that http://www.junghanns.net/asterisk/page14.html |
| 18:29.49 | *** join/#asterisk ctooley ([U2FsdGVkX@199.89.146.18) |
| 18:30.11 | george_ | anyone using gentoo here for *? |
| 18:30.24 | fa | yes, I |
| 18:30.33 | george_ | are you suing MOH? |
| 18:30.33 | ctooley | anyone here sell toll free services? |
| 18:30.52 | Qwell | george_: You can't use the mpg123 ebuild. You'll have to compile it yourself. make mpg123 from the asterisk source dir |
| 18:30.53 | fa | no |
| 18:30.54 | george_ | fa: "using" MOH, that is |
| 18:30.56 | Godsey | is there a correct way to rotate asterisk logs? |
| 18:31.08 | george_ | really? why is that? |
| 18:31.27 | Qwell | because they changed some options, and it returns different stuff now. You need to use 0.59r |
| 18:31.39 | mishehu | if asterisk is using cdr_mysql, why does it still output csv logs? |
| 18:31.41 | george_ | but I have 059s on my suse box. |
| 18:31.53 | file[laptop] | mishehu: because the module is loaded? |
| 18:31.56 | vaewyn | ctooley: nufone.net |
| 18:32.10 | file[laptop] | mishehu: you can have multiple CDR handlers |
| 18:32.17 | fa | george_ i use ZAP |
| 18:32.27 | george_ | Qwell: I'm using mpg123-0.59s-513 on my SuSE box and ti works... |
| 18:32.30 | Qwell | george_: the results are "unexpected". 0.59r is known to work though |
| 18:32.46 | *** join/#asterisk visik7 (~ciao@host174-36.pool80182.interbusiness.it) |
| 18:32.47 | mishehu | file[laptop]: I'd have figured it would be smart and only load cdr_mysql. ;-) guess I'll have to noload the csv one |
| 18:32.52 | Qwell | Which is why they included a "make mpg123" option in the makefile |
| 18:33.13 | Qwell | george_: I recommend unmerging mpg123, and compiling a "proper version" from source |
| 18:33.16 | george_ | Qwell: Okay. I'll build 059r from source. Weould I be better off to convert them all to wav files and use an alternative MOH method. |
| 18:33.21 | george_ | ? |
| 18:33.31 | Qwell | george_: dunno, I'm just relaying what I've seen/heard |
| 18:33.44 | george_ | It's worth a try... |
| 18:33.45 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 18:37.29 | george_ | Qwell: did yu say that the source for mpg123 is included with the asterisk source? |
| 18:37.31 | DaLion | anyone have cdrtool DB fields from billing_customers exl[pantions ? |
| 18:37.48 | Qwell | george_: included, no. But if you "make mpg123", it'll wget the source |
| 18:37.53 | sjaak538 | Godsey use in cli>logger rotate |
| 18:38.04 | george_ | gotcha, thanks! |
| 18:38.15 | *** join/#asterisk George1 (~irc123@24.247.63.62.gha.mi.chartermi.net) |
| 18:38.30 | Qwell | for some reason, it didn't compile for me like that, and I had to go in and do it by hand. ymmv |
| 18:40.23 | fa | I have that " |
| 18:40.24 | fa | exten => 1234/0691761693,1,Wait,1 |
| 18:42.50 | *** join/#asterisk netsurfer (~netsurfer@dreambox.myvnc.com) |
| 18:42.58 | *** part/#asterisk George1 (~irc123@24.247.63.62.gha.mi.chartermi.net) |
| 18:44.41 | fa | anyone use php agi? |
| 18:45.27 | tafazzi | visik7, c'è #asterisk-it per chat in italiano... Se vuoi |
| 18:45.46 | DaLion | seviuo |
| 18:46.00 | DaLion | no mi capiche italiano |
| 18:46.01 | Nugget | tu vuo fa l'americano |
| 18:46.19 | DaLion | jbot: babelfish en it i dont speak italian |
| 18:46.25 | DaLion | hrrh |
| 18:47.16 | tzanger | hahaha |
| 18:47.18 | tafazzi | Sorry it was a message for a possible italian person, to inform him about #asterisk-it a channel for asterisk in italian... Sorry for the wide distribution... |
| 18:47.51 | tafazzi | But I opened an Italian chat room... |
| 18:47.56 | tafazzi | here... |
| 18:48.29 | Nugget | it's unfortunate that irc can't reliably handle non low-bit ascii text. |
| 18:48.30 | tafazzi | jbot: babelfish it en non parlo inglese |
| 18:48.46 | Beirdo | hehe |
| 18:48.59 | Luhiwu | anyone is using chan_h323 for incoming calls? i want to configure the asterisk so calls from one IP goes into context X and from another IP goes into context Y, anyone have done that? |
| 18:49.28 | Beirdo | that bot rules |
| 18:51.33 | Sedorox | jbot: babelfish en it Hello |
| 18:51.40 | Sedorox | hehe |
| 18:51.49 | Qwell | I always thought that meant bye, heh |
| 18:51.59 | Sedorox | jbot: babelfish en it bye |
| 18:52.01 | Sedorox | think its the same |
| 18:52.09 | Sedorox | hmmm |
| 18:52.11 | fa | which is the best protocl, if i am connecting over 512 kbps fro iax phone to asterisk ? |
| 18:52.12 | Damin | Alright.. |
| 18:52.44 | Luhiwu | "ciao" is hello, you may be thinking in spanish "chau" (bye) |
| 18:52.52 | Damin | I've got echocancelwhenbridged=no, but when I conference two Ulaw calls together there is horrible echo. |
| 18:53.04 | Qwell | fa: 512kbit can handle the "heavier" codecs. Mostly, its a give between call quality, and bandwidth used. |
| 18:53.09 | Damin | So the question really is.. Why does everyone reccomend echocancelwhenbridged=no? Seems to me that it should be "yes" |
| 18:53.16 | tafazzi | ciao is confidential... arrivederci is more formal. |
| 18:53.17 | Qwell | fa: basically, test a bunch, and use whichever you think is best for your situation |
| 18:53.33 | Qwell | Don't msg me |
| 18:53.49 | Damin | fa: Use lpc10 |
| 18:54.15 | Damin | fa: Don't message me.. |
| 18:54.18 | Qwell | heh |
| 18:54.20 | Damin | fa: I don't have time to talk to you.. |
| 18:54.20 | fa | hyh ;] |
| 18:54.28 | fa | I must talk with sb.. |
| 18:54.31 | fa | i have a few question |
| 18:54.33 | Qwell | fa: Read your IRC netiquette again |
| 18:55.00 | Qwell | so ask in the channel |
| 18:55.38 | fa | I want to use PHP scripts in AGI.. |
| 18:55.41 | fa | in asterisk |
| 18:55.43 | Sedorox | Question... if two *'s are hooked together via IAX... a phone is connect to A... and the voicemail system is running on B.. .will the phone on A still get the notification that there are messages waiting in its mailbox? |
| 18:55.49 | fa | it's working good? |
| 18:56.12 | Nugget | Damin: I am not certain, but I believe that suggestion is an artifact of earlier bugs which made the echocanceller do bad things with bridged calls. it was intended as a temporary solution until that bug could be resolved. |
| 18:56.12 | fa | and about ast_data, anybody use that, anyone have structure for IAX table in postgresql and some dump of records? |
| 18:56.22 | fa | what about groups of users in IAX2 and SIP? |
| 18:56.23 | Nugget | I don't know if the bug is resolved, though. that's just my recollection of the issue |
| 18:56.37 | Qwell | fa: All of your questions can be answered with the wiki |
| 18:56.41 | Qwell | voip-info.org |
| 18:57.05 | fa | Qwell no. I can;'t find a structure and example record for iax in postgresql. |
| 18:57.12 | fa | and i can't find opinion about ast_data.. |
| 18:57.13 | Zeeek | grep "fa.*wiki" | wc -l ->>> 22345 |
| 18:57.21 | Qwell | Zeeek: Are you serious? |
| 18:57.26 | Zeeek | close |
| 18:57.28 | Zeeek | all day |
| 18:57.28 | Qwell | :p |
| 18:57.43 | Qwell | So, I'm wasting my breath then |
| 18:57.49 | Zeeek | in a word, yes. |
| 18:57.53 | Qwell | fa-- |
| 18:57.53 | Qwell | heh |
| 18:58.04 | *** join/#asterisk netsurfer (~netsurfer@dreambox.myvnc.com) |
| 18:58.05 | tih | Sedorox: No, it won't. |
| 18:58.12 | fa | Qwell what? |
| 18:58.18 | Qwell | nothing |
| 18:58.19 | Zeeek | hi tih |
| 18:58.25 | Zeeek | rough day? |
| 18:58.29 | fa | Qwell tell me ;] sth intersting.. |
| 18:58.37 | tih | Hey, Zeeek -- nope, I've been on holiday for a week. |
| 18:58.42 | Sedorox | damn... |
| 18:58.55 | Zeeek | tih d'you know anything about DSL from an ISP point of view? |
| 18:59.12 | Zeeek | I'll ask my question anyway and anyone who knows can answer |
| 18:59.14 | Damin | Zeeek: Yeah.. what do you need to know? |
| 18:59.21 | tih | Zeeek: I sort of should, since I work for an ISP that does DSL. |
| 18:59.39 | tih | Zeeek: except I start working for a VoIP outfit tomorrow... :-) |
| 18:59.42 | Zeeek | We had a power failure at the office a couple days ago and the digium cards shorted the phone lines until they were restet |
| 18:59.53 | Zeeek | reset |
| 19:00.00 | *** join/#asterisk CCDAS (~spears_da@65.163.100.254) |
| 19:00.18 | tih | Zeeek: that sounds less than polite of them. |
| 19:00.21 | Zeeek | anyway, the DSL connection was broken for more than a day - I was wondering if the DSL provider had equip to sense a short circuit and disconnect? |
| 19:00.44 | Zeeek | I remember being told that the phone co does this in case of a short or did years ago |
| 19:00.49 | netsurfer | probably tripped something |
| 19:01.00 | Zeeek | but where? At the phone co or the DSL prov? |
| 19:01.01 | file | atleast you haven't blown out a line card |
| 19:01.43 | tih | Zeeek: um, the DSL and the phone lines are separate things, so shorting out the phone lines shouldn't make them do anything to your DSL lines. |
| 19:01.43 | netsurfer | phone co |
| 19:01.43 | Zeeek | Yeah the cards looked scray but they came back ok |
| 19:01.43 | file | eventually my telco ran a new phone line for me |
| 19:01.43 | netsurfer | why did they short ? |
| 19:01.43 | Zeeek | well in this country it's more complicated |
| 19:01.43 | file | and voila - all problems fixed |
| 19:01.45 | AsteriskNooB | sorry, i know I asked already but then I had to leave the computer. how do I enter a pause in a dialstring going out ZAP channels? isn't it P? because it's not working and i have 4 of them PPPP trying to give it a 2 or more second lag for the old system to grab a line |
| 19:01.51 | Zeeek | ya see all phones lines go to France Telecom |
| 19:01.59 | Nugget | d'accord! |
| 19:02.20 | Zeeek | then FT "decgroups" them |
| 19:02.26 | Zeeek | routing to the provider |
| 19:02.30 | netsurfer | Zeeek - same here.. they go to BT then BT route the dsl traffic to the isp |
| 19:02.34 | tih | Zeeek: sure, but at the nearest interchange to your site, the phone lines and the DSL lines plug into different types of equipment. |
| 19:02.48 | tih | Zeeek: oh, except when they don't, eh? |
| 19:02.53 | Zeeek | so any answer to my qwest? Because I didn't know who to call |
| 19:03.11 | netsurfer | if ur phone dont work call ur phone co |
| 19:03.15 | netsurfer | if ur dsl dont work call ur isp |
| 19:03.32 | file | check to make sure the phone line is plugged into the phone first |
| 19:03.38 | Nugget | who do you call when your keyboard is dropping letters? |
| 19:03.41 | tih | Zeeek: if your DSL lines are switched by the phone company to your DSL provider, they may indeed be something besides clean, undisturbed, copper. |
| 19:03.43 | Zeeek | yes except that the ISP put me on hold for over a minute then said no one avail - five times |
| 19:03.46 | netsurfer | here, the isp checks their side.. if its ok then contact the phone co |
| 19:04.12 | netsurfer | in the uk u cant report a dsl fault to the phone co. they dont want to know |
| 19:04.18 | tih | Zeeek: in Norway, DSL lines always go to a DSLAM at the nearest exchange. |
| 19:04.35 | tih | netsurfer: same here. |
| 19:04.55 | tih | netsurfer: and its the ISP who must talk to the phone company. |
| 19:05.00 | netsurfer | if the isp cant fix it then THEY call the phone co |
| 19:05.04 | netsurfer | yep |
| 19:05.43 | PakiPenguin | -tih: same here |
| 19:05.49 | *** join/#asterisk jalsot (~tamas@abacus.eworldcom.hu) |
| 19:06.42 | *** join/#asterisk florz (nobody@odnb-d9baa423.pool.mediaWays.net) |
| 19:07.00 | harryvv | Has anyone had any negative feedback from there clients on the spa 841's|? |
| 19:07.10 | *** join/#asterisk FryGuy (fryguy@c-67-174-57-164.client.comcast.net) |
| 19:07.35 | DaLion | guys |
| 19:07.38 | DaLion | from cdrtool |
| 19:07.40 | DaLion | Lookup the billing Profile in cdrtool.billing_customers table in the |
| 19:07.40 | DaLion | <PROTECTED> |
| 19:07.56 | DaLion | can we change to accountid,subs,domain,gateway ? |
| 19:08.13 | DaLion | and is subs the callerid ? |
| 19:08.24 | DaLion | no darn docs on thid |
| 19:08.43 | Zeeek | ok guys thx - the problem is I couldn't reach the ISP |
| 19:09.23 | *** join/#asterisk greg_work (~greg@d221-73-198.commercial.cgocable.net) |
| 19:10.51 | *** join/#asterisk Derkommissar (~Loving@66.64.215.7.nw.nuvox.net) |
| 19:10.55 | Derkommissar | Hello |
| 19:11.25 | file | 'Hello' is overrated, we now say 'Brimoblattingsplat' |
| 19:11.45 | Derkommissar | talpsgnittalbomirB ? |
| 19:11.55 | tzanger | Brimoblattingsplat? |
| 19:11.55 | file | that's acceptable |
| 19:12.18 | Derkommissar | Brimoblattingsplat |
| 19:12.32 | tzanger | oh you mean BrEHmoblattingsplat... dammit learn to spell |
| 19:12.48 | tzanger | I've never seen someone type with an accent before |
| 19:12.55 | file | Russellllllllll |
| 19:13.02 | drumkilla | hey file :) |
| 19:13.08 | file | hi drumkilla |
| 19:14.02 | drumkilla | how's it going? |
| 19:14.20 | file | not too bad, working on some IAX2 stuff |
| 19:14.26 | file | u? |
| 19:14.27 | drumkilla | coding? |
| 19:14.39 | drumkilla | I'm just looking at some email in between classes |
| 19:14.45 | file | indeed coding |
| 19:15.03 | Sedorox | is there a way to link the voicemail systems together? |
| 19:15.11 | drumkilla | file: tell me more! |
| 19:15.21 | file | drumkilla: it's called, Push! |
| 19:15.35 | file | drumkilla: your asterisk box can push/pull data from other asterisk boxes |
| 19:15.39 | file | like... mailbox status! |
| 19:15.43 | Sedorox | oooo |
| 19:15.57 | file | or... CDRs after a native transfer! |
| 19:16.07 | drumkilla | file: what new functionality will that bring? |
| 19:16.16 | freat | anyone have a macro for rolling over to multiple extensions? I'm thinking something that has a variable number of parameters, that uses ifgoto to know if to try the additional extensions... |
| 19:16.26 | file | drumkilla: you can store your voicemail on a remote server, and MWI still works |
| 19:16.33 | drumkilla | fun times |
| 19:16.39 | drumkilla | why not use some management foo? |
| 19:16.42 | Sedorox | file: when do you think you'll have that done :-P |
| 19:16.44 | file | it's going surprisingly well |
| 19:16.48 | file | Sedorox: MWI stuff already works |
| 19:16.55 | Sedorox | across servers? |
| 19:16.58 | file | drumkilla: there's other things you can do with this |
| 19:16.59 | freat | I want to use it to add peoples' cellphones into the mix on their extension. Problem with doing & on dial is if there cellphone is off it goes right to VM |
| 19:16.59 | file | yes |
| 19:17.03 | vaewyn | file file he's our man... if file can't do it... then it shouldn't be done |
| 19:17.04 | Sedorox | in 1.0.3? |
| 19:17.07 | drumkilla | file: I'm just picking on you :) |
| 19:17.09 | file | this is for CVS head |
| 19:17.13 | Sedorox | ah ok |
| 19:17.32 | file | it's verrrrrry cool |
| 19:17.34 | Sedorox | thats one thing I'm really looking for |
| 19:17.41 | file | it's generic too so other stuff can be transmitted |
| 19:18.15 | Sedorox | :) |
| 19:18.29 | file | I'm open for suggestions |
| 19:18.53 | netsurfer | has System been removed as a command ? WARNING[3892]: pbx.c:1294 pbx_extension_helper: No application 'System(/usr/bin/mv etc etc |
| 19:19.07 | Sedorox | yea.. I'm looking at it for a hosting company that has people (support.. sales.. etc..) all over the US and Canada... and we'll have multiple links down to the PSTN links in each country |
| 19:19.10 | george_ | Qwell: yes, taht new mpg123 did the trick on one system. Teh other one still doesn't spit out sound. |
| 19:19.13 | *** join/#asterisk lung (~lung@24-148-96-186.ip.mhcable.com) |
| 19:19.16 | vaewyn | file: remote dialplan update ;P |
| 19:19.38 | Derkommissar | why cant a call join a queue with no valid agents |
| 19:19.40 | vaewyn | switch => on steroids! :} |
| 19:19.41 | Derkommissar | :( |
| 19:19.52 | file | vaewyn: *G* |
| 19:19.53 | drumkilla | vaewyn: that's DUNDi :p |
| 19:19.55 | Beirdo | oh jeez |
| 19:20.03 | file | remote asterisk provisioning and distributed user/peer list |
| 19:20.10 | file | so you configure one asterisk and it propogates to the rest |
| 19:20.16 | Beirdo | I've been looking for a Digium reseller in the Toronto area |
| 19:20.23 | Beirdo | and I found one, and it's someone I know :) |
| 19:20.30 | vaewyn | drumkilla: dundi requires remote lookup... there are many situations where that is not a "good thing" |
| 19:20.46 | Luhiwu | anyone is using chan_h323 for incoming calls? i want to configure the asterisk so calls from one IP goes into context X and from another IP goes into context Y, anyone have done that? |
| 19:21.00 | Sedorox | Beirdo: hehe |
| 19:21.03 | Beirdo | and you'd better believe I'll put money in his pocket |
| 19:21.25 | Zeeek | quick someone with a GS BT100 |
| 19:21.36 | Zeeek | I'm helping somone troubleshoot - what's the default password? |
| 19:21.37 | Sedorox | yea? |
| 19:21.41 | Sedorox | admin |
| 19:21.46 | Zeeek | what I thought, thx |
| 19:21.47 | george_ | anyone have experience w/ MOH? I have a system that starts up mpg123 but I cannot hear it down the line? |
| 19:21.49 | Sedorox | I believe |
| 19:21.51 | Sedorox | let me check |
| 19:21.55 | Zeeek | oh great :) |
| 19:22.23 | Sedorox | errr... maybe I deleted the userguide.. but I believe it is asmin |
| 19:22.24 | Sedorox | admin |
| 19:22.30 | Zeeek | haha |
| 19:22.36 | Beirdo | Oh, and he takes PayPal |
| 19:22.37 | Beirdo | :) |
| 19:22.42 | Sedorox | hehe |
| 19:22.51 | Sedorox | Zeeek: I have two.. pretty nice for the price... |
| 19:23.01 | Zeeek | yes it is admin, it worked |
| 19:23.16 | Sedorox | :) |
| 19:23.26 | Sedorox | http://grandstream.com/Product_Spec.pdf <-- the user guide for it |
| 19:23.29 | Sedorox | very helpful |
| 19:23.42 | Zeeek | I have it but I'm talking to someone and didn't want to look thru it |
| 19:23.49 | Sedorox | errrrrr |
| 19:23.53 | Sedorox | http://grandstream.com/user_manuals/budgetone100.pdf |
| 19:24.03 | Zeeek | yeah I have it on this machine thx |
| 19:24.04 | Sedorox | ok |
| 19:24.13 | Sedorox | kk.. well it is in there :-p |
| 19:24.45 | Zeeek | admion |
| 19:24.48 | Zeeek | admin |
| 19:25.11 | netsurfer | exten => 100,6,System(/bin/mv /home/netsurfer/file1.gsm /var/lib/asterisk/sounds/local/file1.gsm) <-- this command has stopped working since I went from 1.0.5 to cvs HEAD - what replaces "System" ? |
| 19:27.47 | Sedorox | where are the files stored for like... "thank you" and "Welcome" that are used in the demo? |
| 19:28.31 | fa | /var/lib/asterisk - maybe |
| 19:28.41 | Sedorox | don't have that in BSD... had to create it for moh |
| 19:28.49 | fa | who use AGI/(PHP) or postgresql? |
| 19:29.02 | fa | Sedorox meybe try to find by name |
| 19:29.23 | fa | anyone made a callback for cellular phone? |
| 19:29.49 | Sedorox | hmmm... or... we want to create our own welcome menu type thing... how would I play the files.. and what format? wav... mp3? gsm (how would I creat gsm files?) |
| 19:30.01 | Sedorox | fa: I'm just starting out with it so I dunno how to help you :( |
| 19:30.22 | fa | Sedorox but what is your problem? |
| 19:31.15 | Sedorox | well I haven't looked around much for it.. but to create our own menu with custom voice stuff.. like how would I play it.. and does it matter what file-format its in |
| 19:31.22 | netsurfer | Sedorox - /var/lib/asterisk/sounds |
| 19:32.24 | Sedorox | then how do I reference the files? exten => s,1,(ss-welcome) where ss-welcome.mp3 is in sounds? |
| 19:32.38 | *** join/#asterisk Trionnis (buffy@12-203-113-15.client.insightBB.com) |
| 19:34.35 | bjohnson | Beirdo: contact info for the resller? |
| 19:34.47 | Beirdo | http://www.syonex.com/products/shop/digium.html |
| 19:35.22 | Beirdo | I don't think he has a storefront, but you can ask. His address listed is in Unionville |
| 19:36.03 | bjohnson | fa: callback examples in the wiki |
| 19:36.30 | bjohnson | Beirdo: syonex .. I already have that name .. didn't I give it to you? |
| 19:36.48 | Beirdo | I don't think you did, but I don't remember |
| 19:36.49 | fa | bjohnson Can You give a link, i have a problem with find it |
| 19:37.00 | harryvv | Has anyone had any negative feedback from there clients on the spa 841's|? |
| 19:37.00 | Beirdo | I actually know the owner though, that's the sad thing :) |
| 19:37.04 | *** join/#asterisk letherglov (~letherglo@8036aa5e.resnet.ucsd.edu) |
| 19:37.05 | bjohnson | ~docs |
| 19:37.06 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 19:37.22 | bjohnson | fa: ^^ I think on tips and tricks page |
| 19:38.17 | fa | thanks |
| 19:38.23 | fa | on voip-info, right? |
| 19:38.27 | bjohnson | fa: I haven't done it |
| 19:38.30 | bjohnson | yes |
| 19:38.36 | *** join/#asterisk allgood (~allgood@200-101-232-031.fnsce7004.dsl.brasiltelecom.net.br) |
| 19:38.53 | bjohnson | I'm still looking for someone who has a fax/data switch as part of their installation |
| 19:39.14 | allgood | I'm looking for a free and good soft phone... can be IAX or SIP... but must support consulted transfer... can anyone help me onthis? |
| 19:39.25 | Sedorox | algorithmn: x-lite |
| 19:39.28 | Sedorox | errrr |
| 19:39.34 | Sedorox | allgood: X-Lite |
| 19:39.34 | allgood | x-lite doesn't support transfer at all |
| 19:39.38 | Beirdo | you'd better be saluting with all yer fingers :) |
| 19:39.39 | Sedorox | hmmm |
| 19:39.48 | Sedorox | Beirdo: hehe |
| 19:39.57 | george_ | anyone able to offer advice on getting MOH to work? One system is being trouble and I'm not sure why... |
| 19:39.57 | allgood | at least... I couln't make it make a transfer |
| 19:40.15 | Sedorox | I did notice it was greyed out... |
| 19:40.18 | Sedorox | dunno how |
| 19:40.35 | Sedorox | Beirdo: that site is a little expensive |
| 19:40.46 | allgood | Sedorox: can you use transfers with x-lite? |
| 19:40.57 | Beirdo | not really |
| 19:41.00 | Beirdo | that's in CDN $ |
| 19:41.08 | *** join/#asterisk kimosabe (~natt@dsl-200-78-71-49.prod-infinitum.com.mx) |
| 19:41.09 | fa | bjohnson what do you think about it. I am making a system to configure asterix for me. adding iax users etc. What is better to seek thast data by ast_data in database or to parse file ? |
| 19:41.34 | Sedorox | Beirdo: but when I converted it to US.. it came to $107 for a iaxy.. which is $99 on digium's site... |
| 19:41.34 | *** join/#asterisk Tili (~Tili@202-133-65-34-dialup.sat.net.pk) |
| 19:41.42 | Sedorox | allgood: I don't know.. I haven't really tried with transfer |
| 19:41.54 | Beirdo | huh? |
| 19:41.59 | allgood | sedorox, i tried... that buton is disabled! :-D |
| 19:41.59 | Beirdo | it's $100CDN |
| 19:42.03 | Beirdo | plus taxes |
| 19:42.16 | Beirdo | that's cheaper than from digium |
| 19:42.19 | Beirdo | oh wait |
| 19:42.21 | Beirdo | wrong page |
| 19:42.22 | Beirdo | :) |
| 19:42.29 | allgood | Sedorox, I've just tried IAX Phone right now... it supports transfer, but only the blind type... not the consulted type |
| 19:42.40 | Beirdo | $125CDN |
| 19:42.47 | Beirdo | which is pretty close to $99US |
| 19:42.56 | Sedorox | where you getting your price? |
| 19:42.56 | bjohnson | Beirdo: 133 CDN including PST |
| 19:42.57 | *** join/#asterisk aiser (~chatzilla@host238-158.pool8254.interbusiness.it) |
| 19:43.02 | bjohnson | err GST |
| 19:43.09 | Beirdo | 143.75 in Ontario |
| 19:43.13 | aiser | hello to all |
| 19:43.13 | Sedorox | oh yea.. forgot.. that includes the taxes.. so... |
| 19:43.15 | Beirdo | so what? :) |
| 19:43.21 | Sedorox | Beirdo: nm them :-p |
| 19:43.35 | Sedorox | allgood: I'm not sure.. I haven't really messed with softphones that much... |
| 19:43.37 | aiser | I have a question |
| 19:43.51 | allgood | Sedorox, ok... thx for your help... |
| 19:43.53 | Sedorox | just ask :-p |
| 19:43.53 | aiser | someone is working on VoiceXML integration for asterisk ? |
| 19:43.55 | bjohnson | $133.25/1.07=$125 CDN ... $125/1.35=$92.5 USD |
| 19:43.56 | allgood | can anyother help me? |
| 19:44.04 | Beirdo | Sedorox: and it's local. |
| 19:44.09 | Sedorox | Beirdo: yea... |
| 19:44.14 | bjohnson | and no customs broker fees |
| 19:44.17 | Beirdo | bjohnson: ummm. 125*0.80 you mean |
| 19:44.21 | clive- | allgood have you tried iaxcomm? |
| 19:44.24 | Sedorox | Beirdo: thanks for the site... the company I'm helping with is based in canada |
| 19:44.29 | EvlHimeko | cheaper than buy from the states after shipping and getting your anus enlarged from a courier's brokerage charges |
| 19:44.42 | Beirdo | $125CDN = $100US |
| 19:44.43 | Sedorox | yes |
| 19:45.00 | allgood | clive-, not tried... but on feature list I can see only the blind transfer |
| 19:45.03 | Beirdo | :) |
| 19:45.06 | Beirdo | anyways... |
| 19:45.08 | bjohnson | Beirdo: depends on current exchange rate .. I use 1.35 normally .. I don't think we've hit $0.80USD yet |
| 19:45.18 | Sedorox | thats actually when we ordered the grandstream phones.. we had them shipped here (I'm in the US) and then I re-shipped one to canada.. instead of having to do customs twice |
| 19:45.21 | Beirdo | we have been around 0.80US for over a month |
| 19:45.27 | fearnor | hrm |
| 19:45.37 | fearnor | anyone has a LERG that's less than 1 year old? |
| 19:45.40 | Sedorox | I just use xe.com for change |
| 19:45.53 | EvlHimeko | it hit .80 back in october or something |
| 19:46.12 | EvlHimeko | maybe not quite that long ago |
| 19:46.13 | Beirdo | <PROTECTED> |
| 19:46.13 | Beirdo | , 1 Canadian Dollar (CAD) = 0.80684 US Dollar (USD) |
| 19:46.27 | Beirdo | the US$ is sucking rocks, that's why :) |
| 19:46.32 | EvlHimeko | still it's not liek you can get that rate |
| 19:46.32 | Sedorox | lol |
| 19:46.34 | Beirdo | anyways. |
| 19:46.43 | EvlHimeko | that is the interbank rate |
| 19:47.04 | bjohnson | we did a job for a US firm a while back and it got into an argument about what the exchange rate was .. I finally had to explain to my coworker that US uses a different calculation .. so the actual number comparison was meaningless .. they could both be right |
| 19:47.25 | Sedorox | hmmm |
| 19:47.26 | fearnor | there are different rates. |
| 19:47.29 | fearnor | spot rates etc. |
| 19:47.56 | bjohnson | the royal bank site usually has purchase rates and buy back rates listed |
| 19:47.56 | EvlHimeko | buy and sell rates are always different form the interbank |
| 19:48.00 | Beirdo | either way, they will range by about 5% from interbank |
| 19:48.14 | *** join/#asterisk mtvoip (~ircuser@ops-sys-gw.monmouth.com) |
| 19:48.48 | Beirdo | the point being, syonex's prices ain't bad if you are in Canada :) |
| 19:48.58 | Sedorox | Beirdo: yes |
| 19:49.04 | bjohnson | Beirdo: no .. they're good prices |
| 19:49.19 | bjohnson | that's why I had them bookmarked |
| 19:49.22 | Beirdo | :) |
| 19:49.30 | Beirdo | and I'm all for making the owner money |
| 19:49.33 | bjohnson | I thought I gave you that url with a couple others |
| 19:49.45 | Beirdo | I didn't think so, but you may have |
| 19:50.02 | mtvoip | Morning. Is there anyone who can help me with a sip/caller id bug in v1.0.5? |
| 19:50.41 | Beirdo | man does openh323 take forever to compile |
| 19:50.43 | clive- | allgood, I tried to get iaxcomm going on my computer, with little luck |
| 19:51.05 | *** join/#asterisk Mother_ (~mother@93.Red-80-32-127.pooles.rima-tde.net) |
| 19:53.07 | allgood | clive-, used consulted transfer? |
| 19:53.18 | netsurfer | can anyone recommend a gsm codec for converting mp3 files in winxp ? |
| 19:53.22 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@d01m-6-3.d4.club-internet.fr) |
| 19:53.56 | tessier_ | netsurfer: Convert them to wave first |
| 19:54.06 | *** join/#asterisk znoG (gs@200.115.216.109) |
| 19:54.06 | netsurfer | tessier - which type ? |
| 19:54.20 | tessier_ | netsurfer: How many types are there? |
| 19:54.29 | tessier_ | Then use sox to convert the wav to gsm |
| 19:54.29 | netsurfer | Adobe Audition has 4 different types of .wav |
| 19:54.33 | tessier_ | oh geez |
| 19:54.38 | tessier_ | heck if I know. The one everyone uses. |
| 19:54.43 | netsurfer | lmao |
| 19:54.45 | tessier_ | The one asterisk users. |
| 19:54.46 | clive- | allgood, I wannted to test the new jitter buffer stuff in iaxcommm |
| 19:54.53 | george_ | when I convert audio to MP3 for MOH, do I need a specific frequency and otehr options? or will args to mpg123 take care of it? |
| 19:54.54 | netsurfer | maybe sox will deal with them all |
| 19:55.00 | tessier_ | That's the nice thing about standards: So many to choose from! |
| 19:55.18 | *** join/#asterisk ChulJin (~chuljin@65.211.236.166) |
| 19:55.21 | tessier_ | george_: if mpg123 can play it, it should work |
| 19:55.30 | tessier_ | asterisk doesn't care, it just gets a decoded audio stream |
| 19:55.58 | george_ | well, it's on a headless machine and I cannot test it, really... |
| 19:56.27 | george_ | I have audio I ripped with cdparanoia dand encoded with lame. I have ti configured and the mpg123 processes are being started. |
| 19:56.34 | george_ | However, no audio comes down the line... |
| 19:56.37 | Sedorox | if I dial a extention.. how do i get it to play a mp3 file...? |
| 19:57.18 | george_ | Sedorox: playmp3 or musiconhold() should do it. |
| 19:57.32 | Schism | how about streams? |
| 19:57.37 | Schism | like di.fm :-P |
| 19:57.46 | george_ | tessier_: any thoughts on what I should look at? |
| 19:58.00 | george_ | I think there's something on voip-info.org that talks about using streams. |
| 19:58.02 | Schism | press 2 for hard trance :-P |
| 19:58.13 | tessier_ | george_: Did you create a MusicOnHold extension to dial into |
| 19:58.14 | tessier_ | ? |
| 19:58.14 | Schism | press 5 for classical |
| 19:58.22 | tessier_ | pres 6 for african deathmetal bagpipes |
| 19:58.27 | Schism | haha |
| 19:58.29 | Schism | exactly |
| 19:58.36 | george_ | Yes, I have it detect my CXID and it puts me there. I see no errors, but I hear no audio. |
| 19:58.37 | tessier_ | We had a zillion musical genres |
| 19:58.40 | Schism | hahaha |
| 19:58.46 | tessier_ | And african deathmetal bagpipes was the de-facto joke genre |
| 19:58.48 | Beirdo | press 7 for monty python |
| 19:58.49 | *** join/#asterisk eKo1 (~bernd@63.245.57.70) |
| 20:00.11 | Schism | werd |
| 20:00.20 | Sedorox | george_: I'm looking at doing this for menu options.... how would I go about doing it..? |
| 20:01.24 | george_ | are they just going to be put into indefinite hold, or are they going to be waiting? |
| 20:01.29 | george_ | for something? |
| 20:01.38 | terrapen | why do these polycom phones have to be so doggone complicated |
| 20:01.47 | terrapen | there's like five files |
| 20:02.03 | terrapen | no, wait, six or seven |
| 20:02.13 | flewid | sup |
| 20:02.51 | allgood | clive-, now my problem is only the transfer... i liked the "iax phone" but it doesn't do consulted transfers... only the blind ones |
| 20:03.06 | *** join/#asterisk venix (~venix@209.5.255.68) |
| 20:03.25 | flewid | this pa-168s allows me attended transfers |
| 20:03.35 | flewid | but i'm using cvs-head with the new features.conf |
| 20:04.04 | flewid | oh nevermind, it doesn't now :) |
| 20:05.37 | file[laptop] | Yes, I rather like this God fellow. He's very theatrical, you know, a pestilence here, a plague there. Omnipotence. Gotta get me some of that. |
| 20:05.57 | *** join/#asterisk mac_7 (~karsten@d022021.adsl.hansenet.de) |
| 20:06.10 | Sedorox | file: LOL.... love it... |
| 20:06.49 | *** join/#asterisk adnans (~adnans@linuxgoeroe.demon.nl) |
| 20:06.53 | clive- | allgood, maybe the pa168s phone is a good option |
| 20:07.25 | flewid | clive-: the pa-168x based phones are working alright for me so far |
| 20:07.36 | flewid | needs a lot of work but they're just beta firmwares so i can't complain |
| 20:07.43 | flewid | it's nice to have options besides the sometimes flaky iaxy |
| 20:08.22 | file[laptop] | woot IAX2 |
| 20:08.34 | file[laptop] | go IAX2 gooooooooooooooooo |
| 20:08.37 | flewid | i was bored this weekend |
| 20:08.40 | flewid | check this out |
| 20:08.45 | flewid | http://www.nastybits.ca/personal/sounds/ |
| 20:08.47 | flewid | l: guest |
| 20:08.48 | denon | pa-168x? who makes them? |
| 20:08.48 | flewid | p: gu3st |
| 20:09.01 | *** join/#asterisk Syncros (~sysop@noc.routermonkey.net) |
| 20:09.12 | flewid | denon: they are made in asia - that seshu guy from the mailing list sells them in america |
| 20:09.15 | flewid | the atcom phones |
| 20:09.17 | clive- | flewid I am using them , they work well, but I use SIp , havnt tried oax2 yet |
| 20:09.19 | flewid | or netweb-ip301 |
| 20:09.27 | clive- | or yuxin |
| 20:09.30 | wasim | hail oax2! |
| 20:09.34 | flewid | yeah or yuxin |
| 20:09.37 | flewid | and of course the farfon |
| 20:09.38 | flewid | ! |
| 20:10.26 | *** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com) |
| 20:10.34 | flewid | if only sipura would branch out and create iaxura :) |
| 20:10.35 | stevekstevek | clive-: what didn't work? |
| 20:10.46 | clive- | hey stevek :) |
| 20:10.57 | stevekstevek | hola |
| 20:11.14 | clive- | my machine seemd to lose iaxcomm.dll, said it was not locatable...I left it to try tomorrow |
| 20:11.36 | flewid | i wish mastercard would get back to me |
| 20:11.42 | flewid | i want to order my new g4 powerbook :( |
| 20:11.57 | file[laptop] | I ordered my new Dell machine, but it has yet to be approved ;( |
| 20:11.58 | file[laptop] | makes me sad |
| 20:12.15 | flewid | shitty - my gf's aunt is getting a dell shipped here for me to fix up before she takes it home |
| 20:12.22 | flewid | i'd get a pc laptop but those macs are just sooooooooo purty |
| 20:12.26 | flewid | and.. it's a mac :) |
| 20:12.34 | *** join/#asterisk m-00kie (3704558@pcp0010383411pcs.arlngt01.va.comcast.net) |
| 20:12.42 | m-00kie | hello |
| 20:12.44 | m-00kie | sipura site is down? |
| 20:13.15 | flewid | appears so to me as well |
| 20:13.16 | m-00kie | im having trouble with my sipura -- when i plug it into the net, the lights blink 3 fast, 2 slow.. never registers on the server.. any idea? |
| 20:13.48 | *** join/#asterisk jarrod (jarrod@dipole.informationwave.net) |
| 20:13.49 | flewid | file: you get a 64? |
| 20:14.12 | allgood | clive-, where can I find this phone? |
| 20:14.28 | harryvv | mookie, what model? |
| 20:14.32 | doughecka_ | crap |
| 20:14.33 | m-00kie | SPA-1000 |
| 20:14.37 | doughecka_ | sipura's website is down |
| 20:14.39 | doughecka_ | =D |
| 20:14.49 | m-00kie | why '=D' ? |
| 20:15.01 | flewid | allgood: yuu int he states? |
| 20:15.11 | flewid | allgood: that's where i got mine - i'm in canada |
| 20:15.12 | flewid | http://ipphone.eezeephone.com/ |
| 20:15.28 | flewid | model 301 is 1 port, 302 is 2 port, they also have newer models, but no prices listed yet |
| 20:15.36 | harryvv | either there site is down or there backhal fiber link is |
| 20:15.43 | doughecka_ | =) |
| 20:15.45 | allgood | flewid, no... i'm in brazil |
| 20:15.51 | harryvv | flewid where at? |
| 20:15.53 | allgood | flewid, can I buy it online? |
| 20:16.06 | m-00kie | harryvv - any idea what the blinks mean? its 3 fast blinks then 2 slow blinks, repeating cycle |
| 20:16.08 | flewid | allgood: yeah he accepts paypal |
| 20:16.12 | znoG | allgood: colombia 1 - brasil 0 -- what happened there?? :) |
| 20:16.17 | flewid | allgood: http://voip-info.org/wiki-Atcom |
| 20:16.29 | flewid | ^^ i added all the crap i experienced with this phone to that page |
| 20:16.37 | allgood | flewid, I was looking for softfones... but a cheap hardphone can make into my project |
| 20:16.39 | flewid | so you're set to go |
| 20:16.40 | harryvv | mookie, dont know google it |
| 20:16.44 | doughecka_ | which fone is this? |
| 20:16.50 | m-00kie | tried. cant find any info. |
| 20:16.51 | clive- | flewid do you have a site with any decent manual on that phone? |
| 20:16.56 | flewid | doughecka_: pa-168x based |
| 20:17.00 | flewid | clive-: aredfox.com |
| 20:17.05 | flewid | or centrality.com i think |
| 20:17.12 | doughecka_ | ah |
| 20:17.26 | file[laptop] | oh, my, god, BECKY! |
| 20:17.41 | file[laptop] | flewid: yes |
| 20:17.46 | file[laptop] | flewid: 64-bit 3.4GHz Intel P4 with HT |
| 20:18.05 | flewid | file[laptop]: nice, i was going to hold out till the powerbook g5's |
| 20:18.12 | flewid | but now that they upgraded the processors and ram today |
| 20:18.18 | flewid | i can't stand it anymore :) |
| 20:18.23 | allgood | I liked the tin-can phone on that site! wanna one of those |
| 20:18.29 | file[laptop] | it came to... $456.29 USD |
| 20:18.31 | harryvv | :) |
| 20:18.38 | flewid | file: haha uncomparable to mine |
| 20:18.43 | flewid | 4.5k cdn after tax |
| 20:18.52 | file[laptop] | yow'sa |
| 20:19.09 | flewid | 17", 100gb, 1.67ghz, 1gb ram, airport, 8x superdrive |
| 20:19.15 | flewid | panther 10.3 |
| 20:19.23 | file[laptop] | my old box is turning into a server at the school I go to till I graduate |
| 20:19.30 | allgood | flewid, us$ 74,99 is more than the price I get for the budgetone phones |
| 20:19.31 | file[laptop] | then it will become a second workstation |
| 20:19.35 | EvlHimeko | i like laptops to be more portable |
| 20:19.41 | allgood | not buyed any of those yet... but I'll try |
| 20:19.43 | EvlHimeko | liek 10" screens |
| 20:19.47 | flewid | allgood: yeah they are a little more expensive |
| 20:19.48 | tessier_ | I like my girlfriend to be prettier |
| 20:19.56 | flewid | EvlHimeko: yeah, i/m not 100% decided on the 17" yet |
| 20:20.06 | flewid | that might be too bulky for planes and trains and shit |
| 20:20.14 | flewid | 15" |
| 20:20.15 | allgood | flewid, my project is focused on the soft ones... |
| 20:20.18 | modulus_ | <PROTECTED> |
| 20:20.18 | modulus_ | <PROTECTED> |
| 20:20.20 | flewid | allgood: ah |
| 20:20.22 | modulus_ | why is that last hop so lagggy? |
| 20:20.25 | EvlHimeko | i took a 14" over half of japan |
| 20:20.30 | EvlHimeko | to big imo |
| 20:20.33 | EvlHimeko | er too |
| 20:20.34 | flewid | EvlHimeko: yeah ? |
| 20:20.42 | flewid | the only other option is a 12" |
| 20:20.46 | flewid | but that seems just way small |
| 20:20.59 | znoG | eek, did www.nufone.net get bought out by Apache? ;) |
| 20:21.06 | allgood | gotta go... |
| 20:21.08 | allgood | c ya |
| 20:21.31 | EvlHimeko | i like my methos' fujitsu, 10.4" widescreen |
| 20:21.39 | djMax | should I be able to call into * using h323? |
| 20:21.48 | modulus_ | nufone is showing default apache install test page |
| 20:21.55 | modulus_ | on redhat |
| 20:21.56 | modulus_ | eww |
| 20:22.15 | znoG | yup |
| 20:22.20 | znoG | just when i wanted to check my account status |
| 20:22.24 | znoG | balance even |
| 20:22.25 | *** join/#asterisk mrgoby (~mrgoby@pcp05308058pcs.wanarb01.mi.comcast.net) |
| 20:22.36 | modulus_ | maybe they're adding the "options" page |
| 20:22.41 | modulus_ | where it used to say "coming soon" |
| 20:22.49 | EvlHimeko | wxga |
| 20:22.52 | znoG | options.. options are good |
| 20:24.31 | mrgoby | msg jbot seen JerJer |
| 20:24.42 | mrgoby | whoopsite |
| 20:25.06 | ariel_ | djMax, you can if you install h323 on asterisk. |
| 20:25.16 | tessier_ | h323 is evil and asterisk sucks with it |
| 20:25.19 | tessier_ | Avoid it at all costs |
| 20:25.29 | ariel_ | I agree with tessier |
| 20:25.31 | djMax | I'm able to make calls out now, which is a big improvment over my last attempt to do this. |
| 20:25.36 | djMax | But I can't make calls in |
| 20:25.44 | djMax | (I also can't hear the * side yet, but first things first) |
| 20:26.22 | djMax | actually, no, I can hear the * side, not the h323 side |
| 20:26.33 | ManxPower | Don't expect H323 to work thru NAT. |
| 20:26.49 | djMax | yeah, I gave up on that long ago. |
| 20:26.54 | djMax | now they're on the same router even. |
| 20:27.04 | mrgoby | or really at all... h323 is like the cancer of voip |
| 20:27.23 | doughecka_ | lol |
| 20:27.26 | ariel_ | why use h323 when there are so many other good providers out there that use sip and iax2. |
| 20:27.44 | djMax | I am trying to use VoIP to make the most of our Polycom video conferencing system. |
| 20:27.47 | doughecka_ | like, yea |
| 20:27.53 | djMax | bah, I mean "H.323". :) |
| 20:28.08 | doughecka_ | there isnt any diference =) |
| 20:28.14 | Luhiwu | i have to use h.323 here because there are no cheap local providers using SIP or IAX |
| 20:28.18 | Beirdo | ariel_: because work uses H.323 :) |
| 20:28.31 | djMax | and until somebody completely erases h.323 from all the digium and wiki pages, I'm sticking to my hopes that it might work. |
| 20:28.33 | mrgoby | where are you Lihiwu ? |
| 20:28.45 | doughecka_ | djMax: hold on... |
| 20:28.51 | Luhiwu | mrgoby, Argentina |
| 20:29.16 | Luhiwu | i'm having a small problem with chan_h323 |
| 20:29.31 | mrgoby | yeah, central and south america are hard to deal with in that respect... we did a lot of business in mexico and EVERYONE uses h323... |
| 20:29.41 | Luhiwu | i want to configure chan_h323 so calls from one IP goes into context X and from another IP goes into context Y, anyone have done that? |
| 20:29.43 | djMax | what's happening with it? Im definitely no expert, but at least * doesn't crash anymore. |
| 20:29.54 | djMax | h323.conf should let you do that easily |
| 20:30.17 | *** part/#asterisk mrgoby (~mrgoby@pcp05308058pcs.wanarb01.mi.comcast.net) |
| 20:30.29 | djMax | if "h.323 show codecs" doesn't show any, I assume something is wrong? |
| 20:30.42 | Luhiwu | mrgoby, i'm trying to start using IAX2 and offer termination using IAX2 to anyone, but i need h.323 to talk with the rest of the country :) |
| 20:32.41 | djMax | wow, latency on h.323 with two machines right next to each other is not so pleasant. |
| 20:32.46 | file[laptop] | everyone offers termination now |
| 20:32.47 | file[laptop] | even my toaster!!! |
| 20:33.47 | m-00kie | my toaster only offers burnt crumbs :/ |
| 20:34.11 | ariel_ | file[laptop], I don't offer termination yet. |
| 20:34.12 | file[laptop] | awww is it an imitation toaster? |
| 20:34.26 | *** join/#asterisk Delmar (~Delmar@222-152-57-78.adsl.inspire.net.nz) |
| 20:35.20 | mtvoip | Hi. Is there anyone who can help me with a sip/caller id bug in v1.0.5? |
| 20:35.23 | djMax | any thoughts on why I wouldn't hear the H.323 side of a channel? |
| 20:35.50 | *** join/#asterisk jayden (~ircatjerr@65.170.43.34) |
| 20:36.11 | ariel_ | djMax, are the codec's the same? But just guessing I don't use h323 |
| 20:36.20 | PTG123 | Hey i need to play a message while i connect someone to an extensions, how would i do thaT? |
| 20:36.44 | jayden | so, does anybody know if asterisk MOH is asterisk specific or used elsewhere as well? |
| 20:36.59 | djMax | I can't get asterisk to tell me what the codecs in use are (or at least I can't figure out how) |
| 20:37.19 | modulus_ | PTG123, Dial(,,m) |
| 20:37.22 | modulus_ | m = music on hold |
| 20:37.27 | jayden | dj, when you open your console, use more v's |
| 20:37.29 | PTG123 | is that the only way to do it? |
| 20:37.35 | modulus_ | afaik |
| 20:37.45 | buddah | is there some trick to getting a budgetone phone to register with sip? |
| 20:37.49 | PTG123 | modulus_, and does it stop music on hold once call is connected or let it finish out? |
| 20:38.06 | buddah | i did the things it said to on the wiki |
| 20:38.07 | dan2 | markping |
| 20:38.11 | modulus_ | ptg123, mine stops as soon as it's callee picks up |
| 20:38.12 | dan2 | kram: ping |
| 20:38.13 | buddah | but it wont register |
| 20:38.20 | dan2 | can Asterisk listen for sip on multiple ports? |
| 20:38.38 | flewid | modulus_: with ,m in the dialplan , does that mean whenever someone is dialing they hear MOH instead of ringing? |
| 20:38.38 | djMax | NativeFormat/ReadFormat/WriteFormat: are those numbers codec numbers? |
| 20:38.48 | PTG123 | modulus_, yah i have to have message complete |
| 20:39.02 | *** join/#asterisk MAN_Hater (~Too_Rude@adsl-70-240-80-46.dsl.hstntx.swbell.net) |
| 20:39.03 | PTG123 | its to make someone thing person is on the phone before they pick up |
| 20:39.57 | harryvv | When I play a voicemail in a mailbox get this message any idea what it means? Saving message as is |
| 20:39.57 | harryvv | Unable to create lock file: No such file or directory |
| 20:39.57 | harryvv | yay! |
| 20:41.01 | MAN_Hater | waves a happy wave - it looks like ya'll make a nice software product - thank you for the thought that went into this |
| 20:42.50 | *** join/#asterisk randu (~randu@dsl-pppoe-pool1-37.d02.ncshlt.infoave.net) |
| 20:42.56 | tzanger | is anyone having trouble with nufone right int eh last 5 min? I am getting WEIRD status from both switch-1 and switch-2 |
| 20:43.49 | harryvv | tzanger, hears a complaint earlier of there slow ping/tracert times. |
| 20:46.13 | djMax | the polycom seems to want g.722, am i scrod? |
| 20:46.33 | ManxPower | djMax, Huh? Polycom supports several codecs. |
| 20:46.43 | djMax | Polycom Viewstation, not the ip phones |
| 20:46.44 | ManxPower | Oh! Polycom H323. |
| 20:46.52 | djMax | that or g.728 |
| 20:47.13 | randu | Hellow Everyone!!! exten => 707,2,Dial(SIP/${EXT707}@broadvoice,15,tm) when the called person hits # to transfer the call nothing happens. any idea idea why? |
| 20:47.15 | ManxPower | no ulaw or G729? |
| 20:47.27 | djMax | doesn't seem so. |
| 20:47.29 | ManxPower | randu, DTMF problem? |
| 20:47.30 | harryvv | anyone recomend the spa 841 as a starter package for small bizzineses? |
| 20:47.55 | clive- | harry sipura make gppd stuff, so why not |
| 20:47.57 | ManxPower | harryvv, If you don't want PoE or additional switch port, it seems to work well. |
| 20:48.03 | eKo1 | I condemn you to hell HandyTone... |
| 20:48.06 | randu | ManxPower: I will look at that, that is the only thing that I can think of |
| 20:48.07 | eKo1 | Argh. |
| 20:48.14 | ManxPower | The display is not backlit. |
| 20:48.55 | harryvv | Manx, just want to start with them untill I get some more funds together to buy the polygons. BTW what is the Poe and switchport for? sipuras site is down. |
| 20:48.56 | multrix | do somebody know about VoIP phones with gigabit miniswitch (2 ports ) and PoE with gigabit ??? |
| 20:48.58 | ManxPower | ragnar, you want the CALLED person to be able to hit #, right? |
| 20:49.00 | djMax | I see a post from somebody with a g722 implementation, but it's not asterisk specifc, so not sure if that's a long road to nowhere. |
| 20:49.23 | flewid | T = callee can transfer, t = called can transfer |
| 20:49.23 | flewid | :) |
| 20:49.29 | ManxPower | harryvv, PoE = Power over Ethernet, Switch Port: allows you to plug the phone into the ethernet, then the PC into the phone. |
| 20:50.04 | randu | ManxPower: if you were talking to me yes # |
| 20:50.06 | ManxPower | flewid, Not according to the "show application dial" docs. |
| 20:50.14 | tzanger | hmm I need to devise a better way to bounce between providers quickly |
| 20:50.17 | flewid | ManxPower: no? |
| 20:50.21 | flewid | i believe that's what it says on the wiki |
| 20:50.21 | tzanger | maybe some GetDB/PutDB Magic |
| 20:50.26 | ManxPower | randu, So you want the person that the call is going to to be able to transfer the call? |
| 20:50.36 | `Sauron | Randu, dude |
| 20:50.40 | `Sauron | What up |
| 20:50.41 | ManxPower | flewid, The SOURCE is "show application dial" |
| 20:50.44 | randu | ManxPower: yes |
| 20:50.51 | flewid | ManxPower: yes i realize that |
| 20:50.52 | ManxPower | weird, but OK. |
| 20:51.09 | flewid | # t: Allow the called user to transfer the call |
| 20:51.10 | flewid | # T: Allow the calling user to transfer the call |
| 20:51.14 | flewid | http://www.voip-info.org/wiki-Asterisk+cmd+Dial |
| 20:51.16 | randu | ManxPower: if accounting needs to transfer to shipping for example or transfer to a person's cell phone |
| 20:51.18 | flewid | just sayin that's where i saw that |
| 20:51.23 | flewid | sorry to relay information :) |
| 20:51.24 | harryvv | Manx, okay so it needs a external power from the wall to run it. What about the jack part? that was not clear are you saying it is both a pots/etherport capable like that of a ata? |
| 20:51.39 | randu | `Sauron: Still dealing with call transfer issue |
| 20:51.45 | `Sauron | Bummer |
| 20:51.49 | ManxPower | randu, What horrid crappy phones are you using that don't support transfer? |
| 20:51.52 | `Sauron | Any new info on the WiSIP woes? |
| 20:52.09 | silik0n | anyone know a good softphone for OSX? |
| 20:52.18 | ManxPower | harryvv, without a switch port on the phone you need an ethernet jack for the phone and an ethernet jack for the PC |
| 20:52.19 | randu | ManxPower its a Panasonic cordless connected to my POT line |
| 20:52.26 | ManxPower | Lots of new wireing. |
| 20:52.55 | pointer-gaim | who can I bug about merging a patch in mantis into CVS? twisted is away :\ |
| 20:52.58 | ManxPower | randu, Asterisk only handles transfers WITHIN asterisk. |
| 20:53.01 | harryvv | so its not wired for ethernet is what you are sauing |
| 20:53.04 | randu | `Sauron I will call right now and will let you know in 50 minutes :-) |
| 20:53.06 | harryvv | saying |
| 20:53.10 | `Sauron | Laugh. :) |
| 20:53.15 | Delmar | man im still having a wierd echo problem, but now, it only seems to occur on incomming calls. Outgoing calls via the X100P card work great. anyone got any ideas why im getting self-echo on the SIP client only when the call is incomming? |
| 20:53.40 | randu | ManxPower: yea It was my impression that asterisk listens on the line for the # and allows the transfer |
| 20:53.55 | ManxPower | harryvv, Think about this: You bring a phone into an office that has only only ethernet jack and the PC in the office is already using that ehternet jack. Where are you going to plug the phone's ethernet connection into? |
| 20:54.21 | flewid | Delmar: i get about 8 seconds of echo at the beginning of each call in or out on my x100p |
| 20:54.25 | flewid | then it just goes away |
| 20:54.28 | ManxPower | harryvv, You would have to pull another ethernet connectin into that office. |
| 20:54.39 | Schism | ManxPower: most hardphones have a 1 port switch |
| 20:54.39 | randu | or get a router |
| 20:54.52 | ManxPower | If the phone has a 2nd ethernet port, you can plug PC <ethernet> Phone <ethernet> Wall |
| 20:55.00 | Schism | so you can plug your computer into the phone, and plug the phone into the wall |
| 20:55.04 | harryvv | manx the other way is with a hub |
| 20:55.06 | ManxPower | Schism, Only the expensive ones. |
| 20:55.16 | Delmar | flewid, if it was doing that, I could fix it using the echotraining settings. |
| 20:55.17 | Schism | my budgetone isn't expensive, and it has that |
| 20:55.37 | ManxPower | Schism, You can plug your PC into the phone and then the phone into the ethernet? |
| 20:55.38 | flewid | mine just started doing it yesterday for some reason :/ |
| 20:55.40 | harryvv | so are you saying to save the office the cost of the hub hook pc into phone and phone into rj45 jack |
| 20:55.48 | flewid | i've played a bit with echo training, but haven't had too much time to worry |
| 20:55.59 | flewid | it's only my home pbx and my sister and gf aren't complaining so.. |
| 20:56.11 | Schism | harryvv: cost is not as big of a deal as maintenence |
| 20:56.13 | ManxPower | harryvv, Save the cost of a SWITCH (you don't want VoIP on a HUB) and/or save the cost of pulling more cable |
| 20:56.18 | Delmar | flewid, just started doing it out of the blue? u didnt mess with anything.. no software? u didnt plug or unplug anything form the physical line ? |
| 20:56.23 | Schism | mainting network equipment can be expensive |
| 20:56.25 | flewid | Delmar: nope |
| 20:56.28 | harryvv | schism I know :) |
| 20:56.31 | flewid | Delmar: one call it didn't do it, next it did |
| 20:56.32 | flewid | has eer since |
| 20:56.39 | Schism | especialy if you want to have a vlan for just phone traffic |
| 20:56.40 | Delmar | wierd. |
| 20:56.42 | flewid | 8 seconds isn't too bad |
| 20:56.48 | ManxPower | I LIKE the SPA-841. It just has a few limitations that may or may not be an issue depending on what you want/need. |
| 20:56.50 | Schism | 802.1p rules :) |
| 20:56.55 | flewid | i doubt the girls even notice it |
| 20:57.01 | flewid | but i do so it'll get fixed soon :) |
| 20:57.18 | flewid | Delmar: i've read on the mailing list people saying echo just 'appears' about 6 months after they purchased the cards |
| 20:57.18 | harryvv | its more what the customer needs. manx, sold any to some small bussiness? |
| 20:57.18 | Delmar | what kinda echo are u talking about tho.... |
| 20:57.22 | flewid | and i'm right about the 6 month mark |
| 20:57.25 | Delmar | self-echo at the sip client end? |
| 20:57.25 | fa | anyone use php agi? |
| 20:57.47 | flewid | Delmar: this is connected via tdm |
| 20:57.50 | ManxPower | harryvv, I have a 60 phone install coming up in 2 months and we'll prolly go with Polycom |
| 20:57.55 | flewid | haven't tested sip - sec |
| 20:58.01 | Delmar | ah ok |
| 20:58.06 | harryvv | so whats the next step up with a phone that has two extentions Poe and two jack ports? |
| 20:58.11 | randu | fa: I was using it then I took it out. it was giving me errors |
| 20:58.12 | ManxPower | For a 10 phone install we'll prolly go with SPA-841 except for the switchboard |
| 20:58.24 | dan2 | ManxPower: do you know how to extract the calling ip address from a sip phone call |
| 20:58.38 | ManxPower | harryvv, Polycom IP 300 comes to mind but it does not have a microphone for the speakerphone. |
| 20:58.47 | ManxPower | dan2, No. I never cared enough. |
| 20:58.51 | harryvv | thats retarded ;) |
| 20:58.58 | randu | ManxPower: I thought with the t option that asterisk listens on the line for the # and then will do the transfer |
| 20:58.58 | Delmar | The echo issues im having are self-echo at the sip client during a received call via pstn/x100p. Dialing out via the x100p and there is no echo. go figure. |
| 20:59.00 | ManxPower | harryvv, No, it's marketing. |
| 20:59.04 | harryvv | okay |
| 20:59.13 | harryvv | so its a headpiece microphone? |
| 20:59.15 | flewid | Delmar: interesting |
| 20:59.15 | ManxPower | randu, It's supposed to, assuming the call is going thru Asterisk |
| 20:59.21 | flewid | Delmar: yeah, i don't seem to hear anything via sip |
| 20:59.28 | ManxPower | harryvv, It has a HANDSET and a headset jack |
| 20:59.35 | flewid | but if i pick up an analogue handset to an incoming pstn call, i hear 8 seconds of echo, but the caller doesn't. |
| 20:59.55 | randu | ManxPower: ok. |
| 21:00.27 | harryvv | you say no microphone but only can hear the person though the speaker but cannot talk back to them :) |
| 21:00.28 | Delmar | yeah. this echo thing seems to mainly occur on the Asterisk side of the card if you will. |
| 21:00.44 | ManxPower | harryvv, correct. Good for spending long amounts of time on hold. |
| 21:00.52 | `Sauron | randu: t allows CALLED party to transfer |
| 21:00.55 | harryvv | okay....that makes alot of sence |
| 21:01.00 | ManxPower | Echo exists even without Asterisk or SIP. You just can't HEAR it. |
| 21:01.00 | `Sauron | T allows CALLING party to transfer |
| 21:01.25 | Delmar | evilbunny helped me yesterday, and offered a couple of great suggestions which got me to where I am now... but given its echo only for received calls, i might be further from a solution. lol. |
| 21:01.27 | randu | `Sauron: yep and with t specificed asterisk does not respond to the # sign :-( |
| 21:01.41 | `Sauron | Hum, bummer. |
| 21:01.41 | file[laptop] | do you have dtmfmode=inband for broadvoice? |
| 21:01.46 | `Sauron | file: yes |
| 21:02.09 | `Sauron | Look at the lower config on the wiki page |
| 21:02.11 | file[laptop] | I was talking to randu |
| 21:02.19 | randu | file: I will check. |
| 21:02.23 | jarrod | if i match inbound sip extensions from asterisk to ser do you just pipe to t_relay to let it connect to the registered extension? |
| 21:02.28 | *** join/#asterisk zuuluu (klineder@adsl-067-035-113-166.sip.bct.bellsouth.net) |
| 21:02.41 | dan2 | can I have each SIP friend listen on a different port |
| 21:03.53 | randu | file: it is missing on the [broadvoice] in the sip, so I will add and test when I get off the phone. Thanks |
| 21:03.53 | ManxPower | dan2, Why would you want to?? |
| 21:04.12 | ManxPower | randu, Did you tell us you were using Broadvoice?? |
| 21:04.17 | dan2 | ManxPower: because broadsoft software has issues talking to more than one user at the same port |
| 21:04.33 | harryvv | manx, do your customers object to the spa 841 limitations like lack of dual jack ports and no Poe? That is the dual extention version right? What do thay go for. Ohhhh bizzare. www.sipura.com just reverted to voipstore. what the heck! |
| 21:04.36 | dan2 | randu: I'm a broadvoice developer, if you need some help you can wait for me |
| 21:04.40 | randu | ManxPower: Yea last week :-) |
| 21:04.40 | djMax | is there any info on the wiki about integrating codecs? |
| 21:04.52 | `Sauron | Mmm. |
| 21:04.59 | harryvv | Did attacom buy sipura or something? |
| 21:05.03 | randu | dan2: ok. |
| 21:05.09 | ManxPower | randu, Thank you for wasting my time. |
| 21:05.16 | `Sauron | dan2: Careful what you give out. You'll have people /msg'ing your door down... |
| 21:05.19 | ManxPower | dan2, Any provider that only supports ULAW sucks. |
| 21:05.23 | randu | ManxPower: what do you mean? |
| 21:05.35 | dan2 | ManxPower: we support g766 and g729 now, shh |
| 21:05.44 | randu | lol |
| 21:05.46 | stevekstevek | g766! |
| 21:05.49 | dan2 | erm |
| 21:05.51 | dan2 | g26 |
| 21:05.56 | dan2 | g726 |
| 21:05.59 | fearnor | g666 baby |
| 21:06.00 | stevekstevek | g26 is my favorite :) |
| 21:06.01 | ManxPower | randu, By not mentioning broadvoice you wasted my time. |
| 21:06.11 | fearnor | the codec of hell |
| 21:06.16 | `Sauron | ManxPower: Time for your metamucil, grumpy old man. |
| 21:06.19 | ManxPower | dan2, and broadvoice now supports RFC2833 DTMF?? |
| 21:06.32 | dan2 | ManxPower: my boss says so, but the implementation in asterisk is a piece of shit |
| 21:06.33 | ManxPower | `Sauron, You'll feel the same way eventually. |
| 21:06.55 | randu | lol |
| 21:07.19 | `Sauron | Dum di dum |
| 21:07.20 | Schism | when? |
| 21:07.21 | ManxPower | harryvv, The customer has a choice. The polycoms are almost 2x as expensive as the SPA-841. |
| 21:07.52 | randu | ManxPower in the dial statement in my origional message did say that I was using broadvoice. Thanks for your feedback :-) |
| 21:08.01 | djMax | what's confusing to me is why I can hear the asterisk side but not the polycom side... seems to imply the poly can deal with * codec |
| 21:08.03 | `Sauron | Indeed |
| 21:08.15 | `Sauron | ManxPower: broadvoice apparently does rfc2833 |
| 21:08.23 | `Sauron | I just called my cell across it... |
| 21:08.47 | ManxPower | randu, You can ONLY use inband DTMF with ulaw or alaw codec. |
| 21:09.23 | `Sauron | Humm. |
| 21:09.44 | `Sauron | Why'd the config file I saw, say to use ulaw when connecting iax2 to fwd... |
| 21:09.52 | fearnor | boredvoice |
| 21:10.01 | `Sauron | If ulaw is as bad as Manxie claims |
| 21:10.16 | fearnor | ulaw is bad?! |
| 21:10.17 | fearnor | who said. |
| 21:10.19 | silik0n | ulaw isnt bad |
| 21:10.22 | ManxPower | Ulaw has great sound quality |
| 21:10.26 | fearnor | ulaw > * |
| 21:10.31 | ManxPower | ulaw sucks up more bandwidth than any other codec. |
| 21:10.34 | silik0n | only an idiot would say that or someone trying to run it over a 14.4 modem |
| 21:10.35 | jaiger | `Sauron: I think cuz ulaw is a least common denominator thing |
| 21:10.36 | fearnor | well duh |
| 21:10.39 | fearnor | heh |
| 21:10.43 | ManxPower | I said that inband DTMF only works with ulaw and alaw. |
| 21:11.04 | `Sauron | Sigh. |
| 21:11.05 | ManxPower | The bandwidth usage is why I say that any provider that only supports ulaw sucks. |
| 21:11.11 | `Sauron | y'all make up your mind |
| 21:11.18 | `Sauron | Yeah, I figured that out |
| 21:11.28 | stevekstevek | Ahh, found it: G.766 Facsimile demodulation/remodulation for digital circuit multiplication equipment |
| 21:11.30 | fearnor | any provider that only supports ulaw needs to invest into TNTs with cheap codecs ;P |
| 21:11.53 | ManxPower | fearnor, Or just enable GSM and G726 codecs. Neither takes up much CPU. |
| 21:12.03 | fearnor | i'd disagree |
| 21:12.07 | harryvv | Is there a way to send a CallerID msg from asterisk to a windows callerid client that pops up? |
| 21:12.12 | fearnor | its noticeable. |
| 21:12.14 | stevekstevek | G.Sup26 Estimating the signal load margin of FDM wideband amplifier equipment and transmission systems  |
| 21:12.14 | stevekstevek | Red Book Fascicle III.2, page 344 |
| 21:12.21 | ManxPower | iLBC, SpeeX, and G729 are also very good codecs, but suck up a lot of CPU |
| 21:12.44 | silik0n | harryvv: I can code one that'll do that for you hah |
| 21:12.55 | ManxPower | harryvv, Yes. It was announced on the mailing list at one time. |
| 21:13.28 | harryvv | solk haha :) yea not right now. :) but its a nice to have. I guess its for the sheer lazy who does not have the phone near buy |
| 21:14.08 | *** part/#asterisk MAN_Hater (~Too_Rude@adsl-70-240-80-46.dsl.hstntx.swbell.net) |
| 21:14.10 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com) |
| 21:14.23 | *** part/#asterisk ellvis (~ellvis@adsl-data-237.84-47-64.telecom.sk) |
| 21:14.59 | silik0n | i'm actually working on some windows client stuff for asterisk that interfaces to the manager interface for extension status and such |
| 21:15.16 | *** join/#asterisk VoicePulse (~VoicePuls@67.132.43.2) |
| 21:17.01 | dan2 | can I do reverse substring? |
| 21:17.16 | dan2 | from the back |
| 21:17.19 | dan2 | instead of the front |
| 21:17.25 | dan2 | erm |
| 21:17.50 | fa | how to show msn numbers? |
| 21:18.03 | *** join/#asterisk Connor- (~billy@198-144-174-5.knx.tn.nxs.net) |
| 21:18.16 | Connor- | anyone know how to get what speed a network card is at in linux? |
| 21:18.39 | `Sauron | Depends on the hardware |
| 21:18.39 | file | mii-tool might work |
| 21:18.56 | file | as `Sauron said, depends on the hardware... you can try dmesg too |
| 21:19.14 | djMax | are any of these codecs backwards compatible to lower level ones? Still trying to figure out if there's a prayer for 722 or 728 |
| 21:19.27 | vaewyn | mii-tool or mii-diag... they work great |
| 21:19.52 | file | do I want to order food tonight? |
| 21:19.55 | randu | that did it I had left out the dtmfmode=inband |
| 21:20.06 | randu | file: Sure! |
| 21:20.07 | learath | food is good. |
| 21:20.15 | `Sauron | randu: this is the transferring thing? |
| 21:20.16 | Beirdo | <PROTECTED> |
| 21:20.21 | randu | file: why cook if you can afford to order :-) |
| 21:20.28 | randu | `Sauron yep |
| 21:21.14 | file | true... true |
| 21:22.07 | file[laptop] | I just ordered food Saturday though... |
| 21:22.16 | Qwell | doesn't matter |
| 21:22.28 | file[laptop] | and it'll take awhile to get here |
| 21:22.33 | Qwell | go pick it up, heh |
| 21:22.44 | file[laptop] | it would take the same |
| 21:22.48 | Qwell | true |
| 21:23.30 | file[laptop] | nah, waste of money |
| 21:23.36 | Qwell | It would also take a while to cook :p |
| 21:23.43 | randu | `Sauron eat some chips while they food gets there :-) |
| 21:23.45 | file[laptop] | microwave! |
| 21:23.51 | `Sauron | randu: Huh? |
| 21:23.52 | randu | lol |
| 21:23.53 | Qwell | Thats not cooking :P |
| 21:23.58 | file[laptop] | in my world it is |
| 21:24.01 | randu | lol |
| 21:24.04 | Qwell | You and me both... |
| 21:24.09 | randu | `Sauron that was meant for file |
| 21:24.13 | `Sauron | aha |
| 21:24.19 | *** join/#asterisk dca[laptop] (~trillian@sta-207-174-139-178.rockynet.com) |
| 21:24.57 | randu | now I need to find some good code for having the secretary call in and record the main greeting and I should have the system ready for production. |
| 21:25.06 | `Sauron | cool |
| 21:25.49 | doughecka_ | ManxPower: how would I get the key to enable the 4 line thing on the SPA phone |
| 21:26.12 | djMax | can anybody describe codec usage between openh323 and *? i.e. openh323 seems to have its own codec set, is that all that matters on an h323 channel? |
| 21:26.18 | ManxPower | doughecka_, your purchase it. |
| 21:26.33 | *** join/#asterisk robf (~robf@208.188.247.3) |
| 21:26.42 | *** join/#asterisk jjg (tink@216.253.86.223) |
| 21:26.45 | jjg | hello |
| 21:26.45 | ManxPower | contact gsmith@voipsupply.com |
| 21:26.45 | doughecka_ | from? =) |
| 21:26.50 | doughecka_ | ah |
| 21:26.57 | Connor- | Hmm.. mii doesn't see those network cards |
| 21:27.06 | jjg | anyone know about E911 ? |
| 21:27.18 | `Sauron | I know about e911 |
| 21:27.29 | jjg | can * connect to a E911 provider or something? or do i need special expensive equipment? |
| 21:27.32 | `Sauron | [e911] |
| 21:27.33 | `Sauron | exten => 911,1,Playback(no-911-2) |
| 21:27.33 | `Sauron | exten => 911,2,congestion() |
| 21:27.38 | jjg | heh |
| 21:28.20 | dan2 | will this match 1062501766530_xm032 |
| 21:28.26 | dan2 | exten => XXXXXXXXXXXXX_xmXXX,1, .... |
| 21:28.45 | harryvv | Sauron while thats funny its also illegal :) |
| 21:28.49 | jjg | anyone providing 911 or 411 services to voip clients? |
| 21:29.08 | ManxPower | jjg, Most people make sure to have one local PSTN line for 911 |
| 21:29.13 | `Sauron | harryvv: IF they want to call 911, they can pick up the land line phone next to the voip phone |
| 21:29.22 | jjg | Manxpower : gotcha |
| 21:29.29 | dan2 | ManxPower: even if your not paying for local phone service, don't they have to provide you with 911 support? |
| 21:29.37 | `Sauron | harry: my voip provider doesn't yet do 911, so I figured why not tell people |
| 21:29.54 | jjg | but I suppose that doesn't leverage all of the 911 tech, cause the origination will appear as my * box, right? and not the VoIP user |
| 21:30.12 | jjg | bettern nothin for sure |
| 21:30.27 | ManxPower | jjg, If you want real 911 support you have to pay BIG amounts of money. |
| 21:30.29 | *** join/#asterisk r0d3nt|m (RatMan@64.60.114.35) |
| 21:30.39 | jjg | ManxPower : thas what I thought |
| 21:30.59 | harryvv | Sauron mmmm thats not good. |
| 21:31.11 | jjg | is it ok, to call it 911 support ? |
| 21:31.28 | Qwell | "Please hold. Your call will be transferred to 911, just as soon as this 30 page fax is completed." |
| 21:31.33 | jjg | i'm wondering what the liability issues are concerning this. |
| 21:31.33 | *** join/#asterisk jpablo (~jpablo@host-148-244-137-95.block.alestra.net.mx) |
| 21:31.44 | jjg | anyone know of any info portals concerning this issue? |
| 21:33.24 | bjohnson | anyone know if voipjet will support a lower bandwith codec (other than ulaw)? |
| 21:34.25 | doughecka_ | LOL |
| 21:34.28 | tzanger | bjohnson: uh |
| 21:34.40 | tzanger | bjohnson: voipjet should support ulaw/gsm/ilbc/g729... |
| 21:34.52 | tzanger | bjohnson: have you tried it? |
| 21:37.54 | bjohnson | no .. I'd be fried alive at this point if I messed with codecs that didn't work |
| 21:38.16 | tzanger | bjohnson: you have no maintenance window? or a test account? |
| 21:38.41 | doughecka_ | lol |
| 21:38.44 | bjohnson | yeah .. middle of the night maintenance window |
| 21:38.45 | bjohnson | :) |
| 21:38.57 | doughecka_ | say it was the internet that hiccuped |
| 21:39.16 | tzanger | bjohnson: hell use a specific number that you'd call and have it call user@voipjet2 and have that iax.conf entry limit the codec |
| 21:39.45 | bjohnson | yeah I guess so .. didn't think of that approach |
| 21:40.24 | greg_work | anyone use * to receive faxes with rxfax() ? |
| 21:40.31 | bjohnson | if I drop from ulaw, should I try gsm or something else? |
| 21:41.00 | robf | greg_work: I've done it successfully, but it's been a while. Don't use it on a regular basis... |
| 21:41.39 | jjg | robf : do you mean that you don't use it on a regular basis, or no one should |
| 21:41.46 | greg_work | i thought i had it working, but the faxes didnt come out properly, only the top bit of the page |
| 21:41.54 | robf | I don't -- I'm not making any recommendations... |
| 21:41.57 | *** join/#asterisk scubasteve (~steve@office37.neonova.net) |
| 21:42.05 | robf | I just know it *can* work... |
| 21:42.22 | scubasteve | Anyone got a recent SPA841 firmware? Sipura appears to have disappeared from the net... |
| 21:42.29 | scubasteve | Just had 2 show up in the mail. |
| 21:42.34 | tzanger | greg_work: yup |
| 21:42.38 | tzanger | use it every day |
| 21:42.46 | tzanger | between 30 and 50 faxes a day |
| 21:43.05 | *** join/#asterisk ZX81 (matt@222-152-158-141.jetstream.xtra.co.nz) |
| 21:43.09 | *** join/#asterisk visionv (~sonye@208.239.206.195) |
| 21:43.19 | robf | tzanger: what version of asterisk/spandsp/app_rxfax are you using? |
| 21:43.32 | tzanger | robf: CVS HEAD and 0.0.1pre6 I think |
| 21:43.46 | greg_work | scubasteve, i downloaded the supposedly newer firmware than I had, but when I ran it i ended up with the same version that was on there to start with, though it did say 'succesful'. i dunno if it's just printed wrong on their site or if they bundled the wrong version |
| 21:43.51 | robf | tzanger: istr some lag on the spandsp/app_rxfax side... |
| 21:43.59 | tzanger | what do you mean |
| 21:43.59 | greg_work | tzafrir, i was trying to get it with 1.0.5 stable |
| 21:44.00 | scubasteve | greg_work: Classy. |
| 21:44.33 | scubasteve | greg_work: I'm less than thrilled with the fit/finish on these. I opened one phone up and every key sticks... the line in use LED's aren't lined up with the holes... |
| 21:44.42 | visionv | can someone help me please? I need to know how I would hook up a t-1 to my asterisk machine and also hook up 24 analog phones. Does anyone know what card(s) I need? |
| 21:44.52 | scubasteve | greg_work: Am afraid to open the 2nd. |
| 21:45.05 | tzanger | visionv: are you willing ot do some basic research? |
| 21:45.08 | greg_work | scubasteve, mine are fine in that respect. my biggest complaint is the speaker phone is kind of crappy |
| 21:45.21 | *** join/#asterisk jskcr (~jskcr@jskcr.user) |
| 21:45.24 | ZX81 | who would I contact to get good rates on millions of minutes to Europe? |
| 21:45.25 | greg_work | rather, the microphone. but i haven't really played with it a lot yet |
| 21:45.25 | ManxPower | visionv, You need a T-1/E-1 card and a channel bank. |
| 21:45.32 | scubasteve | greg_work: Haven't even plugged it in yet :-) |
| 21:45.53 | visionv | tzanger: sure. I have a 4 port fxs/fxo running now. I am trying to figue out what equipement and cards I need. |
| 21:46.12 | greg_work | how are the LED's misaligned anyways? they're flush with the black plastic on mine |
| 21:46.13 | tzanger | visionv: as ManxPower mentioned, you want a T1 card and a channel bank |
| 21:46.29 | tzanger | the nomenclature on channel banks is identical to the TDM card so to hook up phones to it, you want FXS ports |
| 21:46.43 | tzanger | they can be had for about $200-$250 for 24 ports from ebay |
| 21:46.48 | tzanger | and a T100P is $495 from digium |
| 21:46.57 | tzanger | I have a number of these running -- they work fine |
| 21:47.09 | scubasteve | greg_work: Not centered in the hole. |
| 21:47.17 | robf | tzanger: what kind do you have? |
| 21:47.31 | visionv | hmm, I thought there was a digium product that took the place of the channel bank |
| 21:47.32 | scubasteve | greg_work: Might be able to pull them into place with a pick.. otherwise might need to crack it open and line them up. |
| 21:48.00 | scubasteve | greg_work: Also convinced there's a bucket of sand or a fishing weight epoxied to the inside of the case.... 841 is way too heavy for what it is. |
| 21:48.14 | visionv | tzanger: Thanks! |
| 21:48.18 | tzanger | robf: I use the Carrier Access Access Bank I for FXS or the Adit600 for FXS/FXO combinations |
| 21:48.34 | bjohnson | I haven't seen chan banks for that cheap .. usually around $500 looks like a deal for a loaded one |
| 21:48.36 | tzanger | do *NOT* use ABI or ABII for FXO, they don't have functional far-end disconnect supervision |
| 21:48.37 | greg_work | yeah, they are decently heavy |
| 21:48.39 | greg_work | thats a good thing though |
| 21:48.40 | zuuluu | can anyone tell me how to set a t100p to no use intenal sync source? |
| 21:48.47 | scubasteve | greg_work: I think the button sticking issue is because the pads are too wobbly (lateral) and get stuck under the hole if you don't push them straight down.. |
| 21:48.49 | tzanger | bjohnson: I get them all the time on ebay for about $250 for 24-port FXS ABI |
| 21:48.57 | tzanger | and significantly cheaper sometimes |
| 21:49.16 | scubasteve | greg_work: For $89 shipped, if it works... cool. This is not the "enterprise business class" phone they tout it to be. |
| 21:49.23 | *** join/#asterisk DaGrim (~jason@dagrim.user) |
| 21:49.26 | greg_work | the handset cords kinda suck though. i've had mine sitting on my desk for like 5 days with the cord dangling down, and its already stretched to almost twice its shipping size :p |
| 21:49.29 | DaGrim | Hey all. |
| 21:49.55 | scubasteve | greg_work: Yeah, noticed that too.. cord is crummy. Hell, it's all screwed up in the stock photos on their website too! |
| 21:50.02 | greg_work | i can't get any buttons to stick |
| 21:50.17 | robf | My CB is an Adtran TA750 -- all FXS but one card (4-ports FXO) |
| 21:50.26 | jjg | tzanger : any particular model for the Carrier Access ? |
| 21:50.53 | greg_work | tzanger: have you used the rxfax with stable? have there been changes to it in HEAD? |
| 21:51.03 | jjg | robf : i was thinking about one of those ... how do you like it? are you using a T100P ? |
| 21:51.03 | robf | pretty happy with it -- better form factor than the CA models |
| 21:51.06 | DaGrim | Im getting this twice when on boot of my box itself.. is this maybe why my codecs have been sucking so bad?: /sbin/ldconfig /usr/local/lib/libspeex.so.1 is not a symbolic link |
| 21:51.07 | tzanger | jjg Access Bank I or II (1 or 2 T1) for FXS, Adit600 for FXS/FXO combinations or FXO only |
| 21:51.12 | robf | jjg: T100P, yes |
| 21:51.13 | scubasteve | greg_work: Strange. I've seen a ton of complaints about buttons and RFI issues... Will have to take out the dual band ham radio gear tonight and see :) |
| 21:51.21 | tzanger | but really if you're doing more than about 12 FXO it's likely cheaper to get a ct1 or PRI from the telco |
| 21:51.23 | jjg | tzanger : looking for FXS only |
| 21:51.31 | vaewyn | Ok guys... so which is the cheapest current Cisco phone with SIP available? |
| 21:51.38 | DaGrim | I use only speex mind you |
| 21:51.41 | fa | how can i get in phpagi a callerid number? |
| 21:51.47 | tzanger | vaewyn: that's teh Cisco ElCheapo... only available in Mexico |
| 21:51.49 | greg_work | tzanger: have you ever had problems with faxes coming out messed up (ie, the top 1" of the page, then just a big scramble that could possibly be the rest)? |
| 21:51.53 | robf | jjg: it was difficult finding (on ebay) an FXO card for it, and not cheap. Originally came full of FXS |
| 21:52.01 | scubasteve | vaewyn: 7960, about $250-$260 us with power adapter. |
| 21:52.04 | vaewyn | tzanger: hehehe... :} |
| 21:52.07 | jjg | robf : are you using cat5 or cat2 for the interconnect? |
| 21:52.08 | scubasteve | 7940 might be slightly cheaper. |
| 21:52.14 | tzanger | greg_work: yup - -frame slip -- make sure you have enough power in your app_rxfax machine and a very clean line |
| 21:52.20 | vaewyn | scubasteve: thanks for the info |
| 21:52.35 | robf | jjg: cat5 cable, but custom-wired... |
| 21:52.38 | scubasteve | greg_work: Wonder if the firmware with the 4 line appearances is keyed to the phone or if it can be "shared" ... |
| 21:52.59 | jjg | robf : is it simple to configure with *? |
| 21:53.01 | jaiger | robf: I have the TA750 too and agree it was tricky to find an FXO card for it |
| 21:53.01 | scubasteve | vaewyn: I belive voipsupply.com is running a special on 7960's right now (no affiliation, just remembered seein' it) |
| 21:53.07 | greg_work | tzanger: like powersupply you mean? i'm using a TDM400P,.. don't have power hooked into the molex connecter on it though -- should i? |
| 21:53.12 | jaiger | the FXS cards are a dime a dozen |
| 21:53.25 | greg_work | scubasteve: i'm not sure how they do that |
| 21:53.29 | tzanger | greg_work: no as in horsepower |
| 21:53.30 | jjg | I was hoping to find a how to on using a TA750,t100p, and * server but can't find anything |
| 21:53.46 | bjohnson | tzanger: do most chan banks also need a patch panel to run the cable to the phones or do some have the modular jacks right in the chan bank? |
| 21:53.48 | tzanger | greg_work: and I have had very spotty issues with RECEIVING faxes to fax machine connected to TDM400P... I can send out of it just fine |
| 21:53.50 | jaiger | jjg: I found it easy once I made my own cable |
| 21:53.55 | robf | jjg: the only problem I had was that, for whatever reason, the rj45 plugs on BOTH ends have to be pushed in really tight, and jiggle loose just by wiggling the cable a little bit. |
| 21:54.05 | greg_work | tzanger: oh .. i should, i think it's like a 2.8ghz athlon |
| 21:54.07 | tzanger | bjohnson: all I have seen use AMP D50 connectors so you need a patch cable to go to BIX or S66 or whatever you use |
| 21:54.10 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 21:54.12 | robf | Once you've got a good connection, though, it's as simple as any other zap channel, I would say... |
| 21:54.16 | jjg | jjg : jaiger : ok, thanks |
| 21:54.29 | jaiger | jjg: config is via RS232 |
| 21:54.48 | tzanger | greg_work: yes hook up that molex connector -- it helps the card keep stable loop current to the phones its servicing |
| 21:54.49 | jaiger | robf: I have echo problems with mine, do you? |
| 21:54.49 | robf | jjg: right -- I made my own 'harmonica'... |
| 21:54.51 | jjg | robg : jaiger : does the TA750 have the slots for the rj11s? or does it need a patch panel? |
| 21:55.05 | robf | jaiger: sometimes, perhaps... not often... |
| 21:55.11 | greg_work | tzanger, i'm only using fxo though |
| 21:55.14 | jaiger | jjg: I bought a patch panel, it has RJ45 for T1 and D50 for analog |
| 21:55.18 | vaewyn | ok... who has the most different phones on hand for testing? (wondering on recommendations for testing) |
| 21:55.23 | jjg | robf : 'harmonica' ? |
| 21:55.24 | tzanger | greg_work: ahh well in that case no it doesn't really matter |
| 21:56.00 | robf | jjg: telco-slang for a telco-50 to broken-out rj11 or rj45 cable... |
| 21:56.28 | robf | http://www.l-com.com/sdex/H76.JSP |
| 21:56.47 | robf | see that url for harmonica example |
| 21:56.48 | jaiger | jjg: I bought from phonegeeks.com ... http://www.phonegeeks.com/noname1.html |
| 21:57.11 | jjg | robf : great, thanks |
| 21:57.19 | file[laptop] | woot microwave |
| 21:57.26 | fearnor | adit600 is the bst. |
| 21:57.28 | fearnor | best |
| 21:57.58 | greg_work | i bought a 96-port RJ45 patch panel (BIX connections) on ebay for $60 with shipping |
| 21:58.08 | jjg | robf : jaiger : so i'm guessing my parts list has grown to 1 TA750, 1 24 port harmonica or patch panel , * server , 1 T100P card ...that sound bout right? |
| 21:58.20 | Mother_ | hi all |
| 21:58.30 | jaiger | jjg: cabling |
| 21:58.38 | robf | jjg: haven't seen all of the conversation, but that sounds like a pretty good setup |
| 21:58.42 | tzanger | yeah I just order that stuff through our local telco :-) |
| 21:58.51 | Mother_ | is there a way that * can turn callerid into a name from a local directory, which is then sent to the SIP phones instead of the number? |
| 21:59.00 | tzanger | Mother_: of course |
| 21:59.05 | Mother_ | I imagined :D |
| 21:59.09 | tzanger | asterisk can do pretty much anything within reason |
| 21:59.16 | tzanger | you can use DBGet if you like |
| 21:59.19 | tzanger | or even something more standard |
| 21:59.24 | Mother_ | I've had a look in the wiki but obviously I'm looking in the wrong place |
| 21:59.35 | Mother_ | OK, what would be more standard? |
| 21:59.45 | ManxPower | The problem is that users won't log out of their phone. |
| 21:59.48 | greg_work | Mother_, well, how is your local directory stored? |
| 22:00.19 | jjg | jaiger : when you've had echo issues with your TA750, is it enough of a problem that I should not consider a TA750 for an office environment? |
| 22:00.19 | Mother_ | greg_work: it's not yet, I'm open to whatever method ends up working best, in terms of management & flexibility and callerid |
| 22:01.00 | jjg | robf : ok, thanks a LOT for the info |
| 22:01.07 | Mother_ | so far I've got the basic * with voicemail etc. working fine, SIP phones are working fine, and want to explore further things |
| 22:01.26 | Mother_ | I managed to solve the hangup problems on the FXOs too |
| 22:01.32 | jaiger | jjg: I'm getting an echo canceller for our office as we speak |
| 22:01.44 | greg_work | well, decide on that first I guess. you can use *'s internal db (DBget to get the numbers and SetCallerIDName() to set the name), or something like a mysql database to store everything (web-based app to manage it, AGI to grab the number and give it to *).. LDAP (which * can access directly).. . |
| 22:01.47 | jaiger | jjg: I have no idea if it's the TA750 causing the echo |
| 22:01.52 | ManxPower | Mother_, You took out bustdetect=yes and/or callprogress=yes? |
| 22:01.55 | jjg | jaiger : how much that gonna run ya? |
| 22:01.58 | jjg | jaiger : ok |
| 22:02.36 | jaiger | jjg: my research indicates all hybrids can source echo, the TA750 is just one point of entry |
| 22:02.40 | *** join/#asterisk Duckbizkit (~Duckbizki@24.240.243.142) |
| 22:02.49 | Duckbizkit | is there a way to define a class of silent hold music |
| 22:02.54 | Mother_ | ManxPower: nope, I changed the settings in wctdm.c to the ones for Spain instead of FCC, recompiled and it seems to work |
| 22:02.58 | jaiger | I figured I'd nip it in the bud and just get the canceller |
| 22:03.01 | jjg | jaiger : that is a bit over my head to understand |
| 22:03.06 | jaiger | jjg: it is annoying enough though |
| 22:03.09 | Mother_ | I cannot rule out blind luck either, but in any case it's fixed :) |
| 22:03.39 | jjg | jaiger : ok, good to know. cause i've got several other locations interested in having the same solution if it works well |
| 22:03.43 | *** join/#asterisk RaYmAn-Bx (user@213.237.12.147.adsl.vby.tiscali.dk) |
| 22:03.46 | jaiger | jjg: by hybrid, I mean the circuit that converts from digital to analog |
| 22:04.03 | Mother_ | greg_work: thanks for those suggestions, I'll look into them |
| 22:04.04 | jjg | jaiger : ok |
| 22:04.27 | jaiger | jjg: essentially my connection to PSTN, at my TA750 |
| 22:05.14 | Duckbizkit | Mother_, any ideas? |
| 22:05.22 | jjg | jaiger : so you think the issue may be mitigated if I'm only doing outbound via IP network? |
| 22:05.25 | fa | how to set cidname from php script by agi? |
| 22:05.44 | Mother_ | Duckbizkit: as for silent hold music? record an audio file with silence :) |
| 22:06.17 | jaiger | jjg: I would think so |
| 22:06.28 | jaiger | jjg: of course YMMV |
| 22:06.41 | *** join/#asterisk CoderCR (~creyna@66.240.200.105) |
| 22:06.42 | CoderCR | hello all |
| 22:07.27 | Duckbizkit | heh, easy enough i guess |
| 22:08.05 | Mother_ | most sound editors allow you to insert silence one way or another, create an empty file, insert xx seconds of silence and repeat for as long as you need |
| 22:08.38 | ManxPower | asterisk-sounds has silent .gsm files of varying lengths. |
| 22:08.57 | Mother_ | well even easier then :) |
| 22:09.05 | *** join/#asterisk santiago (~santiago@201.245.167.72) |
| 22:09.10 | jjg | jaiger : thanks for the info |
| 22:09.15 | fa | anyone know how to include in setcidname a variable returnet by agi? |
| 22:09.20 | *** join/#asterisk oferlin (~oferlin@ALille-251-1-5-20.w82-127.abo.wanadoo.fr) |
| 22:09.43 | *** part/#asterisk CoderCR (~creyna@66.240.200.105) |
| 22:09.44 | jjg | anyone successfully using a TA750 for outbound via IP network only? |
| 22:10.04 | oferlin | Hi, is there any french guy here ? |
| 22:10.51 | labo | feeling lonely ? |
| 22:11.01 | jjg | heh |
| 22:11.04 | Mother_ | haha |
| 22:11.04 | oferlin | mmm ?! |
| 22:11.56 | oferlin | Hi, is there any english guy who speaks english as a french ;-) ? |
| 22:12.58 | robf | who said they were running the latest (or so) rxfax? |
| 22:13.02 | stevekstevek | I can type with a french accent. |
| 22:13.04 | robf | it won't compile for me... |
| 22:13.11 | oferlin | hehe |
| 22:13.19 | eKo1 | Parle français. |
| 22:13.23 | oferlin | great ! |
| 22:13.28 | Mother_ | any oppinions on the Zyxel 2000W? |
| 22:14.14 | robf | ugh... callerid changes... |
| 22:14.21 | fa | ? |
| 22:16.45 | ManxPower | Mother_, None of the 78 google results from the mailing list archives were helpful? |
| 22:17.15 | *** join/#asterisk fafnir (~hello@tdds-gw.Moscow.gldn.net) |
| 22:17.26 | denon | ManxPower: of course not |
| 22:17.32 | Mother_ | as for the Zyxel? |
| 22:17.44 | oferlin | i am looking for sample design to buid an ipbx with asterisk for replacing an alcatel pabx on isdn french network |
| 22:17.45 | Mother_ | sorry, I hadn't looked... |
| 22:17.52 | ManxPower | Mother_, Perhaps you should. |
| 22:18.01 | denon | ~google ManxPower evil evil sob |
| 22:18.11 | denon | wow, no results. |
| 22:18.13 | denon | huh. |
| 22:18.47 | ManxPower | well duh! |
| 22:19.10 | Mother_ | just felt like getting some live oppinions, I've been googling for two weeks and my eyes hurt |
| 22:19.27 | denon | ~google ManxPower satan |
| 22:19.31 | ManxPower | Mother_, with site:lists.digium.com as one of your search terms? |
| 22:19.48 | Mother_ | ManxPower: yes, but not on the zyxel |
| 22:19.51 | denon | man, getting 6k/s from trend micro .. thats just wrong |
| 22:20.00 | file | denon! you're speaking |
| 22:20.07 | Mother_ | as I say, I've been googling tons of stuff as for example the hangup problem |
| 22:20.16 | denon | kinda |
| 22:20.21 | Mother_ | reading the wiki, etc. |
| 22:21.45 | *** join/#asterisk dca[laptop] (~trillian@sta-207-174-139-178.rockynet.com) |
| 22:21.46 | vaewyn | anyone had a chance to play with a gxp-2000 from GS yet? |
| 22:22.06 | denon | unless somethin has changed recently, GS == the suck |
| 22:22.19 | denon | wish they didnt .. they're nice n cheap .. but .. |
| 22:22.41 | vaewyn | Mine doesn't suck... other than the # only callerid |
| 22:22.53 | vaewyn | but I did put a sheet of lead in the bottom of it to add a little weight |
| 22:22.54 | vaewyn | :} |
| 22:23.00 | denon | well, their suckage seems to be when used in heavier production |
| 22:23.17 | denon | not heavier like lead ..... |
| 22:23.25 | vaewyn | umm... I'm on the phone 90% of the day |
| 22:23.31 | denon | yeah, but how many phones? |
| 22:23.36 | vaewyn | that is pretty heavy production use i would say |
| 22:23.41 | vaewyn | 15 on this subnet |
| 22:23.49 | eKo1 | Say, do all contexts where calls are made need a t extensions? |
| 22:24.09 | *** part/#asterisk djin (~marius@gridfox.xs4all.nl) |
| 22:24.10 | tessier_ | eKo1: If you want to handle timeouts, yes |
| 22:24.21 | vaewyn | I do prefer the IP-300s we have but... the GS arn't that bad... and they support ilbc :P |
| 22:24.27 | *** join/#asterisk MykDee (~mdavenpo@proxy-sjc-2.cisco.com) |
| 22:24.33 | denon | and gsm, dont they? |
| 22:24.37 | vaewyn | yep |
| 22:24.45 | denon | yeah, I'd like the IP300s to have gsm and ilbc |
| 22:24.51 | denon | that'd be a snazzy phone |
| 22:24.59 | denon | throw in IAX and it'd be a no-brainer .. well, the 500 would be anyway |
| 22:25.00 | vaewyn | but I prefer ilbc... cause the low bandwidth uses are areas we have high p[acket drop rates (wireless or LRE) |
| 22:25.13 | vaewyn | 500s are nice also |
| 22:25.16 | denon | dunno, we run mostly ulaw |
| 22:25.19 | *** join/#asterisk kimosabe (~natt@dsl-200-78-71-49.prod-infinitum.com.mx) |
| 22:25.27 | denon | dont really care much about bandwidth .. |
| 22:25.29 | Delmar | beh. I give up. this echo thing is so bad I might just have to abort the entire idea. |
| 22:25.33 | vaewyn | we run totally ulaw on campus |
| 22:25.36 | jjg | i'm having trouble getting my X100P to ring for more than one time...does this extensions.conf entry look ok : |
| 22:25.37 | jjg | exten => s,1,Answer |
| 22:25.37 | jjg | exten => s,2,Dial(${HOMEPHONE}|15|r) |
| 22:25.39 | jjg | ? |
| 22:25.48 | vaewyn | but for LRE and wireless ilbc only way to go if you can |
| 22:26.13 | Delmar | ilbc is nice. lower bandwidth but pretty fair quality audio. |
| 22:26.29 | denon | why LRE? |
| 22:26.29 | vaewyn | yep... and handles dropped packets the best |
| 22:26.38 | denon | seems like fiber's generally cheaper for cross-campus runs |
| 22:26.44 | jpablo | hi, is there any one to compile the zaptel driver into the kernel (ie. not as modules) ? |
| 22:26.44 | ManxPower | Why do an answer?????? |
| 22:26.45 | denon | or rather, a better use of funds |
| 22:26.47 | Delmar | has anyone played around with the hacked g729 codec vs the proper licensed one? |
| 22:26.51 | ManxPower | And stop using "r" |
| 22:26.55 | jjg | ManxPower : ok |
| 22:27.03 | jjg | so no answer needed and stop using r |
| 22:27.10 | Delmar | lol. |
| 22:27.16 | Delmar | whats the r for anyway? |
| 22:27.20 | vaewyn | denon: Our appartments are from the 1930s... rewiring of any form is never going to happen |
| 22:27.24 | jjg | someone else created the conf file |
| 22:27.25 | vaewyn | hence... LRE |
| 22:27.27 | jjg | :( |
| 22:27.43 | denon | oh, so its LRE over pots cable? |
| 22:27.47 | denon | good grief |
| 22:27.48 | vaewyn | yep |
| 22:27.57 | denon | I thought you were just doing long reach over cat5 for distance |
| 22:28.02 | ManxPower | Delmar, "r" means provide ringing sound to the caller even when they should hear something different like BUSY or "The number you have called has been disconnected" Asterisk will provide ringing sound to the caller BY DEFAULT. |
| 22:28.16 | vaewyn | only option... buildings are conrete with LOTS of rewire in them... hence wireless isn't even an option |
| 22:28.25 | denon | vaewyn: 802.11g :) |
| 22:28.42 | vaewyn | denon: not an option... I can't pentrate 1 wall... let alone 10+ |
| 22:28.50 | denon | good grief |
| 22:28.52 | denon | tear the place down |
| 22:28.59 | denon | quit trying to put networking in the projects!! <G> |
| 22:29.03 | vaewyn | heh... LRE is cheaper :P |
| 22:29.25 | vaewyn | They are nice... but they were built to last and that REALLY dampens any running of cables |
| 22:29.25 | Delmar | cheers ManxPower. |
| 22:29.37 | Delmar | ManxPower so .. why remove the "answer" ? |
| 22:29.37 | vaewyn | heck... they can barely take the electrical load on their current electric system |
| 22:29.41 | *** part/#asterisk MykDee (~mdavenpo@proxy-sjc-2.cisco.com) |
| 22:29.54 | denon | ghetto.. :) |
| 22:30.18 | vaewyn | denon: They really are nice places... way better than any of the apartments in town |
| 22:30.33 | Bentley | hi all, i've got a quad T1 card that stopped working on reboot (flashing red lights) and I see these in dmesg: "wct4xxp: Setting yellow alarm on span 1" .. anyone know what this means? |
| 22:30.44 | vaewyn | 600sqft for a single.. |
| 22:31.49 | Delmar | ManxPower, if you remove the "Answer" ... does that mean that the line wont be picked up unless the device (homephone in that case) picks up? |
| 22:33.19 | ManxPower | Screen Name Not Available |
| 22:33.19 | ManxPower | <PROTECTED> |
| 22:33.19 | ManxPower | Sorry, AsteriskPBX is already in use. |
| 22:33.19 | ManxPower | Please try again or let us suggest a Screen Name for you. |
| 22:33.21 | ManxPower | LOL! |
| 22:33.39 | ManxPower | Delmar, Correct. Unless you send the call to voicemail or something. |
| 22:34.03 | Qwell | What is the passthrough port on the x100p for exactly? |
| 22:34.05 | Delmar | so lets say you had a scenario where you wanted to operate some standard phones as well as an Asterisk box with some SIP devices... the phones would ring, and asterisk would detect the ringing and make the sip phones ring.. but wouldnt actually pickup the call... |
| 22:34.21 | Qwell | Does that phone still ring, until * picks up, or something? |
| 22:34.33 | Delmar | ManxPower, thats handy to know. |
| 22:35.06 | vaewyn | Qwell: ignore it... it is an evil vestige left from the fact the X100P was a modem |
| 22:35.13 | Qwell | ahh |
| 22:35.15 | Delmar | ManxPower, how would I make several SIP phones all ring, and all be able to pickup and answer the incomming call.. ? |
| 22:35.45 | Qwell | I guess I should buy an FXS card now, or something. How's the IAXy? |
| 22:35.57 | ManxPower | Delmar, Dial(SIP/fred&SIP/john) ; The gay couple |
| 22:36.04 | Delmar | LOL |
| 22:36.09 | Mother_ | har har |
| 22:36.27 | *** join/#asterisk kietlak (~kietlak@11-mo3-6.acn.waw.pl) |
| 22:36.44 | Mother_ | Delmar: check out extensions.conf at http://www.loligo.com/asterisk/current/ |
| 22:36.51 | Mother_ | it has some nice examples on all this |
| 22:38.07 | Mother_ | also look at sip.conf and zapata.conf to get the complete picture |
| 22:38.23 | Delmar | ManxPower, im having some trouble with calerID not being passed to my grandstream phone. It seems to pass it to say.. an xlite sip client ... any trick to doing this? |
| 22:38.30 | file | oh ManxPower dear, how are you? |
| 22:39.26 | Delmar | Mother_ hey thanks. thats going to help me quite a bit. cheers. |
| 22:39.34 | *** join/#asterisk gfivealive (~gandres@atlnga1-ar4-4-33-029-076.atlnga1.dsl-verizon.net) |
| 22:39.39 | denon | is it really necessary to call him dear? |
| 22:39.42 | denon | kinda freaks me out <G> |
| 22:40.10 | file | denon: awwwww |
| 22:40.15 | file | how cute! |
| 22:40.30 | *** mode/#asterisk [+mi] by denon |
| 22:40.33 | denon | ack |
| 22:40.37 | *** mode/#asterisk [-mi] by denon |
| 22:40.43 | denon | friggin .. hrm |
| 22:40.58 | file | now now! |
| 22:41.01 | file | be nice |
| 22:41.15 | denon | I was being nice .. just trying to mute you <G> |
| 22:41.41 | denon | oh alright .. |
| 22:41.42 | denon | :) |
| 22:42.04 | *** join/#asterisk charles___ (~charles@64.35.168.55) |
| 22:43.07 | Delmar | ok.. does anyone in here have ANY idea what the HELL could be causing errors on incomming calls that generate messages like... zt_handle_event: Ring/Off-hook in strange state 6 on channel 1... etc? it does this 4 times, and messes everything up when the call is coming in. |
| 22:43.27 | charles___ | hey |
| 22:43.50 | charles___ | what do you guys recommend for server based on experience: Xeon or AMD 64 ? |
| 22:43.53 | Mother_ | Delmar: I've seen that strange state message only once, and it didn't affect the call |
| 22:44.04 | denon | charles___: xeon |
| 22:44.11 | stevekstevek | sounds like a Ring or Off-hook event was found when it wasn't expected, eh.. |
| 22:44.11 | charles___ | need to handle 4 e1's |
| 22:44.15 | charles___ | fully loaded |
| 22:44.17 | denon | charles___: smp xeon |
| 22:44.34 | denon | though it varies on what you're doing with em |
| 22:44.40 | charles___ | denon amd 64 doesn't handle it ? |
| 22:44.48 | denon | amd64 is fine, but xeon is better |
| 22:44.57 | denon | and more procs is better |
| 22:45.01 | *** join/#asterisk Frantic (~ab@TechnologicPartners35.dsl.concentric.net) |
| 22:45.02 | charles___ | going to compress the tdm to G723 and send over SIP |
| 22:45.03 | denon | as its very well threaded |
| 22:45.22 | denon | you mean E1 to alaw then? |
| 22:45.31 | Frantic | hi guys- anyone using snom here? having poblem with snom reporting busy when one line is on hold |
| 22:45.34 | Frantic | any idea? |
| 22:45.48 | charles___ | e1 allaw transcodec to G723 over IP |
| 22:45.55 | Delmar | Mother_ well, i noticed it was worst when I was messing with the rxgain and thought i had it beat but now its back again... with avengance.... as if i don't have enuf problems with echo and other crap. lol. |
| 22:46.01 | file | asterisk won't transcode that there G723 ya know |
| 22:46.01 | silik0n | y0 denon check you pms |
| 22:46.04 | algorithmn | has anyone ever tried to hook up a data T1 to a TE405p or equivalent zap driven device?? |
| 22:46.09 | gfivealive | i'm new to asterik. are there any good tutorials on how and what you need to setup a small office system based on VoIP. |
| 22:46.12 | Luhiwu | anyone knows a pastebin not broken? pastebin.ca gives me database errors |
| 22:46.17 | Frantic | <algorithmn> I did |
| 22:46.23 | algorithmn | gfivealive: yah, give me a moment |
| 22:46.25 | Frantic | <algorithmn> T100p |
| 22:46.26 | file | Luhiwu: pastebin.com? |
| 22:46.34 | algorithmn | Frantic: i got some questions for you |
| 22:46.39 | gfivealive | great! thx. |
| 22:46.52 | Frantic | <algorithmn> sure |
| 22:46.57 | Delmar | when calls come in.. i just get this... Feb 1 11:44:37 WARNING[27205]: chan_zap.c:3465 zt_handle_event: Ring/Off-hook in strange state 6 on channel 1 |
| 22:46.59 | Delmar | about 4 times |
| 22:47.09 | Luhiwu | file, thanks |
| 22:47.13 | Juggie | bad signaling? |
| 22:47.15 | Delmar | and it stops ringing the SIP device/s, then starts again... |
| 22:47.23 | Frantic | anyone using snom 220? |
| 22:47.27 | Luhiwu | anyone can help with some DTMF problems with * 1.0.5 and SIP? http://pastebin.com/235915 |
| 22:47.29 | Mother_ | Delmar: try to power on/off the box, just in case |
| 22:47.35 | algorithmn | gfivealive: "the asterisk handbook" "hitchhikers's guide to asterisk" |
| 22:47.43 | algorithmn | google will show you the rest |
| 22:47.44 | Delmar | Mother_ hard reset the cards u think? |
| 22:48.00 | Mother_ | yes, power down the PC and back up |
| 22:48.02 | file | Luhiwu: use rfc2833, cause the info type your stuff is using asterisk doesn't support |
| 22:48.06 | Mother_ | as I say, just in case |
| 22:48.07 | algorithmn | Frantic: firstly, is there anything that i should know before i start asking? |
| 22:48.08 | gfivealive | thanks for your help! i'm sure I'll need more l8r. |
| 22:48.17 | Delmar | Mother_ couldnt hurt. its on its way down now. |
| 22:48.19 | algorithmn | feel free to private message me |
| 22:48.27 | Frantic | <algorithmn> free up a weekend for that |
| 22:48.31 | algorithmn | lol |
| 22:48.39 | Delmar | hrm |
| 22:48.45 | jjg | can someone paste in a working extensions.conf entry that gets their x100p to answer the phone? |
| 22:48.45 | Frantic | <algorithmn> siriously- you'll need to recompile the kernel with HDLC support |
| 22:48.51 | algorithmn | ohhh |
| 22:49.02 | algorithmn | not what im wanting to hear |
| 22:49.03 | Delmar | jig i could.. but i just shut down the box for a minute lol |
| 22:49.03 | Luhiwu | file, thanks a lot |
| 22:49.12 | Frantic | <algorithmn> then, change something in the zconfig.h (uncomment the net part) |
| 22:49.13 | Mother_ | jjg: look here http://www.loligo.com/asterisk/current/ |
| 22:49.20 | algorithmn | actually... im running fedora core 2 n i think it supports it?? |
| 22:49.27 | jjg | ok, thanks |
| 22:49.32 | Mother_ | just make sure you understand how he's configuring the context |
| 22:49.32 | Frantic | <algorithmn> i did it on RH9 |
| 22:49.42 | algorithmn | 9 didn't have hdlc? |
| 22:49.48 | file | as I wait for stuff to transfer between boxes so I can testtttttt my codeeeeeeeeeee |
| 22:49.49 | Delmar | Mother_ do u think .. now that its off for a moment.. that it would be a good idea to shove it onto a nice UPS with power filtering? |
| 22:49.55 | algorithmn | kernel level support... i think im seeing the problem |
| 22:50.02 | Frantic | no- it did not |
| 22:50.12 | Mother_ | Delmar: I have all my boxes through UPS by default |
| 22:50.18 | Frantic | <algorithmn> i'm totaly new to linux |
| 22:50.20 | *** join/#asterisk JunK-Y (~junky@modemcable144.95-37-24.mc.videotron.ca) |
| 22:50.24 | Mother_ | last thing I want is the power company screwing them up :) |
| 22:50.24 | Frantic | <algorithmn> but i finally did it |
| 22:50.29 | algorithmn | what a feeling |
| 22:50.37 | algorithmn | thats what makes computing worth whle |
| 22:50.43 | algorithmn | while.. the self satisfication.. |
| 22:51.14 | *** join/#asterisk kippi (fc@cpc4-hatf3-6-0-cust243.lutn.cable.ntl.com) |
| 22:51.15 | kippi | hey |
| 22:51.19 | Frantic | <algorithmn> better yet- the T1 i have is a mix- data and voice |
| 22:51.20 | algorithmn | Frantic: i just put in the order for pri t1 n data t1 and am about to order the 405p |
| 22:51.25 | Delmar | Mother_ was just wondering if i should or not... i mean.. UPS's can cause issues... signwave / squarewave etc... but then...PC's have nice switchmode PSU's with all sorts of their own filtering... |
| 22:51.30 | Delmar | should be good. |
| 22:51.30 | algorithmn | you are the person i was looking to talk too |
| 22:51.51 | Frantic | <algorithmn> i had a lot of sh*t with that |
| 22:51.53 | algorithmn | seriously... partial must be harder then what i'm trying |
| 22:52.05 | Mother_ | Delmar: use good quality UPS, I use rackmount APC |
| 22:52.09 | algorithmn | what was the biggest problem... other then not slaving to the linux kernel |
| 22:52.27 | Frantic | the kernel was the biggest issue |
| 22:52.34 | Mother_ | they are useful too as they can signal the boxes when there are problems, so you can email/pager/gracefully shutdown etc |
| 22:52.41 | Frantic | after that you just need the right scripts and that's it |
| 22:52.43 | algorithmn | mmm... all i need is a bottle of wine and a glass... i'll have the computers |
| 22:52.45 | Frantic | i can help you with that |
| 22:52.52 | algorithmn | i would appreciate it |
| 22:53.02 | algorithmn | agi at all? |
| 22:53.09 | Frantic | msg me for my email |
| 22:53.13 | Delmar | Mother_ yup. |
| 22:53.14 | Frantic | no agi |
| 22:53.39 | Delmar | Mother_ hey something I might mention... when I load the zaptel module, the crc_ccitt module loads as well. |
| 22:53.48 | Delmar | crc_ccitt 1632 1 zaptel |
| 22:53.58 | Delmar | should it be doing that? |
| 22:54.08 | Frantic | snom users? |
| 22:54.08 | Mother_ | I haven't got a clue to be honest :D |
| 22:54.49 | Delmar | anyone else get a module called crc_ccitt loading when the zaptel module is loaded? |
| 22:55.15 | kippi | i am Grandstream HandyTone 286 VoIP Adapter but i can't seem to get it work, how do i get asterisk to see the box and give it an exenstion number? |
| 22:56.46 | Luhiwu | is there any way to force rfc2833 to the other side? i'm receiving INFO messages for DTMF tones, i've put dtmfmode=rfc2833 in my sip.conf but i don't have control to the other side... |
| 22:59.35 | Delmar | Mother_ no go. |
| 22:59.35 | Delmar | still does Feb 1 11:58:41 WARNING[1520]: chan_zap.c:3465 zt_handle_event: Ring/Off-hook in strange state 6 on channel 1 |
| 22:59.38 | Delmar | does that 4 times... then ... |
| 22:59.38 | Delmar | == Spawn extension (fromfxo, s, 1) exited non-zero on 'Zap/1-1' |
| 22:59.38 | Delmar | <PROTECTED> |
| 22:59.50 | Delmar | and while its doing that.. the line is still ringing.... |
| 22:59.52 | *** part/#asterisk santiago (~santiago@201.245.167.72) |
| 22:59.56 | Delmar | and it will repeat that again.. |
| 23:01.17 | *** part/#asterisk cbachman (~cbachman@129.105.7.250) |
| 23:01.25 | Mother_ | hmmmm I can't think of anything then, really |
| 23:01.32 | redder86 | JerJer: around? |
| 23:01.33 | Mother_ | it's not something that I've come across |
| 23:01.42 | Delmar | yep. |
| 23:02.11 | Delmar | it actually works better if I get Asterisk to answer the line within the first ring or so. |
| 23:03.12 | ManxPower | Delmar, you don't have somethig like ringmaster/distinctive ring? |
| 23:03.20 | ManxPower | Don't enable callprogress either |
| 23:03.29 | Delmar | ah ok. |
| 23:03.29 | redder86 | Does anyone else experience that the outset of a VoIP call is more likely to be corrupted than later during the call? I.e., the "ring" sounds by the far end get garbled, but 1 or 2 seconds after the callee answers the problem goes away? |
| 23:03.32 | Delmar | ill try that. |
| 23:04.16 | greg_work | are there any ports i need open in order to accept incoming calls via iax2? |
| 23:05.30 | visionv | tzanger: thanks for the earlier help, I do have another question if you are up to it :) |
| 23:05.49 | visionv | brb |
| 23:06.04 | *** join/#asterisk sivana (~richard@209.91.159.221) |
| 23:06.16 | Delmar | Mother_ thats my zapata.conf .. anything in there that catches your eye? |
| 23:06.48 | sivana | ~seen sixtel |
| 23:06.49 | jbot | sixtel <sixtel@sixTel.iax.cc> was last seen on IRC in channel #asterisk, 25d 17h 48m 4s ago, saying: 'no such host, not in sip.conf right'. |
| 23:06.49 | PTG123 | anyone here use the Queue() command, we are having problems getitng it to work.. says requires an arguement |
| 23:06.50 | Qwell | greg_work: 4569 for iax2, 5036 for iax, both udp |
| 23:07.28 | greg_work | whats 5060/udp and 5038/tcp ? |
| 23:07.38 | Qwell | 5060 is sip |
| 23:07.38 | greg_work | oh, 5038 is manager interface |
| 23:08.33 | Duckbizkit | Mother_, you got a sec? |
| 23:09.00 | Delmar | im gonna run and grab a coffee and something to shove in my pie hole. afk a little. |
| 23:09.05 | greg_work | ah, cool, theres a firewall page on voip-info.org .. i looked for "ports" but there were too many results :p |
| 23:10.56 | *** join/#asterisk ctooley ([U2FsdGVkX@199.89.146.18) |
| 23:12.45 | `Sauron | What the heck |
| 23:13.10 | `Sauron | when trying to call an international number across my spa-1001, it gives me fast busy, but * never sees the attempted dial |
| 23:13.28 | *** join/#asterisk mutombo (mutomb_@pD9E2A58E.dip.t-dialin.net) |
| 23:13.30 | pointer-gaim | `Sauron: sounds like a dialplan issue |
| 23:13.51 | `Sauron | Hrmph. |
| 23:14.12 | `Sauron | On the sipura, though. I don't even see an attempted call in * |
| 23:14.14 | `Sauron | Grr. |
| 23:15.12 | *** join/#asterisk zuuluu (klineder@adsl-067-035-113-166.sip.bct.bellsouth.net) |
| 23:15.13 | greg_work | the sipura has a dialplan option |
| 23:15.21 | pointer-gaim | `Sauron: yup...on the sipura |
| 23:15.26 | `Sauron | I'm looking at it now |
| 23:15.26 | *** join/#asterisk cf (fc@cpc4-hatf3-6-0-cust243.lutn.cable.ntl.com) |
| 23:15.36 | zuuluu | where can i change the hunt sequence for outbound to "bottom to top"? |
| 23:15.39 | greg_work | i just put mine to x. |
| 23:15.46 | `Sauron | It looks plain enough |
| 23:15.50 | `Sauron | (*xx|#xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxx|011xxxxxx.) |
| 23:15.51 | greg_work | as i'd rather have * deal with it, no sense in having two |
| 23:15.57 | zuuluu | or any link with hunt sequence info for asterisk...i dont see anything buy hunt groups which is diff |
| 23:15.58 | `Sauron | Hum |
| 23:16.00 | mutombo | evening |
| 23:16.47 | *** join/#asterisk hans (fugalh@falcon.fugal.net) |
| 23:17.07 | greg_work | actually it needs to be x.|*x. or *-prefix numbers won't work |
| 23:17.12 | mutombo | i ask myself if its possible to reroute a call to another phone via commandline or the agi? |
| 23:17.26 | `Sauron | (x.|*x.|#x.) |
| 23:17.31 | `Sauron | That's what I'm setting it to |
| 23:17.48 | `Sauron | mutombo: I was wondering that too |
| 23:17.54 | greg_work | yeah thats probably good. i dont use #x. for anything right now .. but who knows |
| 23:18.13 | mutombo | so i hold an existing call and then switch it to another phone via an webinterface |
| 23:18.30 | *** join/#asterisk Nugget (nugget@dazed.slacker.com) |
| 23:18.32 | visionv | Got a question: how would I put 1000 customers on a asterisk computer. I mean, is there some kind of switch (like a ethernet switch) that wouls tie to the NIC ? |
| 23:18.33 | hans | is there a secret to getting asterisk to listen for inbound iax calls? I tailored iax.conf and reloaded / restarted, but nothing |
| 23:18.38 | `Sauron | damn |
| 23:18.39 | greg_work | mutombo: theres a flash-based panel that can do it .. so yes |
| 23:18.41 | `Sauron | still busy |
| 23:19.18 | greg_work | visionv, what are you talking about? 1000 phones on the LAN? |
| 23:19.24 | mutombo | flash-based? |
| 23:20.07 | visionv | greg_work: well, I can put a tdm100p and channel bank together and handle 24 users at a time right? |
| 23:20.12 | greg_work | mutombo, asternic.org |
| 23:20.20 | visionv | but how would I grow that to 1000 |
| 23:20.34 | greg_work | visionv, more channel banks. sip gateways. voip phones. |
| 23:20.59 | mutombo | greg_work thx |
| 23:21.41 | fa | anybody know how to make a callback on zaphfc? |
| 23:21.42 | visionv | greg_work: more more channel banks and up to 3 tdm100p's sound ok. voip phones would plug into what? |
| 23:22.14 | Delmar | ok so for future reference... callprogress=no is a good way to fix the incoming call issues I was having. |
| 23:22.23 | Delmar | but I still havent fixed this damn echo :(. |
| 23:22.27 | *** join/#asterisk tensai (~tensai@207.141.37.66) |
| 23:22.33 | ManxPower | There is no such thing as a TDM100P! |
| 23:22.50 | visionv | sorry, t100p |
| 23:22.59 | ManxPower | Delmar, I advocate renaming callprogress=yes to randomlyfuckupmycalls=yes. |
| 23:23.03 | greg_work | visionv, ethernet |
| 23:23.03 | `Sauron | Grr. |
| 23:23.07 | ManxPower | Why not just put in a T400P? |
| 23:23.18 | Mother_ | hahaha |
| 23:23.27 | visionv | greg_work: with an ethernet switch right? |
| 23:23.31 | Qwell | ManxPower: Can I assume that those don't exist either? heh |
| 23:23.36 | ManxPower | Delmar, echocancel=yes and echotraining=yes didn't help? |
| 23:23.38 | greg_work | visionv, yes |
| 23:23.57 | greg_work | visionv, or over an internet connection |
| 23:24.01 | ManxPower | Delmar, Have you tried echotraining=600 or 800? |
| 23:24.01 | greg_work | visionv, or with wifi |
| 23:24.11 | visionv | and no other way with analog phones other than channel banks right? |
| 23:24.11 | greg_work | visionv, this is the power of voip and * |
| 23:24.24 | Delmar | ManxPower not really. lets give that a shot. |
| 23:24.28 | *** join/#asterisk McKillroy (~mckillroy@L0954P06.dipool.highway.telekom.at) |
| 23:24.37 | McKillroy | Hello ! |
| 23:24.46 | ManxPower | visionv, If you want up to 8 analog ports you can use TDM400P cards w/FXS modules. |
| 23:24.49 | McKillroy | Did anyone of you ever tried this: http://www.soft.uni-linz.ac.at/_wiki/tiki-index.php?page=ProjectBluezHandsfree |
| 23:25.09 | `Sauron | greg_work: I set my dialplan as before, but it's still giving fast busy |
| 23:25.11 | `Sauron | any idea why? |
| 23:25.14 | greg_work | visionv, sure there is. the sipura SPA-2000 for example is a SIP device (that physically plugs into ethernet) that gives you two fxs ports. there are bigger SIP gateways too |
| 23:25.22 | Delmar | ManxPower ok just to recap... this is a self-echo at the SIP client that is almost not even there when calls are placed outgoing, but is a royal pain in the ear when incomming calls are answered.. and it doesnt seem to echotrain and go away... |
| 23:25.38 | visionv | greg_work & ManxPower: Thanks !!! |
| 23:27.02 | visionv | brb |
| 23:27.03 | Mother_ | McKillroy: are you still insisting on doing that 'el cheapo' GSM adapter? |
| 23:27.13 | McKillroy | Yepp. |
| 23:27.24 | greg_work | visionv, thats one of the nice things about voip. assuming your network is setup right (the * server is accessable from the internet), I could take the SIP phone sitting on my desk home, plug it in, and I would have an office extension exactly as if i was sitting in the office |
| 23:27.32 | Mother_ | you'll have to write a connector to take the audio from the phone to something * can handle |
| 23:27.37 | McKillroy | I want it. But I'm afraid its either expensive or complicated. |
| 23:28.08 | greg_work | `Sauron, for everything you dial? what did you change? and does * see it trying to call? |
| 23:28.09 | Mother_ | McKillroy: there are tons of GSM/PSTN line simulators out there |
| 23:28.27 | Mother_ | I have a company that manufactures one about 10 minutes from where I live |
| 23:28.31 | McKillroy | I found a GSM to PSTN adapter for 515 euro ... still too much |
| 23:28.36 | Delmar | greg_work, thats what im trying to get working at some point but my testing so far has had wierd results... get this... |
| 23:28.38 | `Sauron | greg: apparently only for int'l calls |
| 23:28.55 | `Sauron | greg: I changed sipura's dialplan to (*x.|#x.|x.) |
| 23:29.16 | *** part/#asterisk hans (fugalh@falcon.fugal.net) |
| 23:29.41 | `Sauron | Ah, blah. Gotta go again. |
| 23:29.41 | Mother_ | McKillroy: this one made next door goes for 278 euros |
| 23:29.42 | Delmar | greg_work, scenario is ... xlite ===> linux router (NAT) ===> Internet ===> Route (NAT) ===> Asterisk. this works MINT. calls both ways, the works...... |
| 23:29.56 | McKillroy | Mother_ : Linky ?? |
| 23:30.04 | McKillroy | Mother_ :Do you know if the product of www.phonelabs.com could be a solution ? |
| 23:30.11 | Delmar | greg_work, replace xlite with a budgetone 102.....and it wont even login. |
| 23:30.47 | Mother_ | McKillroy: http://www.grupohilmon.com/ it's the SISCOM thingy |
| 23:30.55 | greg_work | Delmar, i haven't started playing with NAT stuff at all yet. i just gave my * box an internet ip |
| 23:30.57 | greg_work | :) |
| 23:31.01 | Delmar | if it was a perminant need to have that end connected to me.. I would run asterisk on his end, tie them with IAX and be done with it. |
| 23:31.31 | Delmar | but what im really after.. is to make sure i can .. grab my phone.. go someplace.. plug into internet.. point it to an stun server.. and away I go. |
| 23:32.14 | Delmar | greg_work, yep. im going to put my * box on a public real soon.. to overcome all my issues... i just dont trust shitty D-Link DSL-300's. they dont seem to reconnect if there is a power issue or such like. |
| 23:32.22 | Mother_ | McKillroy: the phonelabs device would be try-and-see, it probably would work as it also simulates a line, but things like caller ID etc. are more doubtful |
| 23:32.34 | Delmar | greg_work, and i disliked the whole.. PPP via an Alcatel ..... |
| 23:32.47 | Delmar | greg_work, i may have to buy a new DSL modem for the task. |
| 23:33.18 | Duckbizkit | hey, what's that site to post configs to an IRC channel |
| 23:33.24 | McKillroy | Mother_ : Caller ID does not matter so much as the price ... its all for private use only, and I have no Ex-GFs terrorizing me ...;) |
| 23:33.25 | Qwell | ~pastebin |
| 23:33.26 | jbot | pastebin is, like, a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 23:33.30 | Delmar | greg_work, so yep. NAT is a big pooo. I dont understand why more phone and ata manufacturers dont move to IAX. |
| 23:33.33 | Duckbizkit | thx |
| 23:33.46 | Delmar | surely grandstream could upgrade their phones in a software update to do IAX.. and NAT wouldnt be an issue. |
| 23:34.35 | Delmar | ok so im gonna play with that idea ManxPower had.. lets see if that does anything. |
| 23:34.55 | Delmar | echotraining=600 or 800 right? |
| 23:35.13 | Duckbizkit | well i've got a problem with queues....we have two servers with the same CVS checkout, same configs....just one works and the other hangs up on the call instead |
| 23:35.14 | greg_work | Delmar, i think they are, just slowly |
| 23:35.20 | Duckbizkit | log here: http://pastebin.ca/5055 |
| 23:35.34 | greg_work | SIP is a standard, IAX2 is a *-proprietary protocol (although it IS open..) |
| 23:35.47 | Delmar | right. well i hope they get into it soon. |
| 23:35.54 | wolfson | iax also does not seem to like packet loss AT ALL |
| 23:35.55 | *** part/#asterisk yogurt2ungue (~charlie@203-132-126-200.fibertel.com.ar) |
| 23:35.58 | Duckbizkit | and i've tried calling Queue() without any args other than the queue name, no help |
| 23:36.18 | Delmar | wolfson, that would be more a codec issue rather than the protocol itself .. no? |
| 23:36.19 | McKillroy | Mother_ : Just a stupid question: PSTN = ?? same like POTS ? |
| 23:36.29 | Qwell | McKillroy: pretty much |
| 23:36.32 | Delmar | PSTN is more or less POTS. |
| 23:36.42 | wolfson | delmar: no idea, but I've noticed that after heavy packet loss, the call is never normal again |
| 23:37.00 | McKillroy | Qwell: I need to read stuff as I see ... |
| 23:37.05 | Mother_ | hehe |
| 23:37.09 | Delmar | wolfson, heh. sounds like what my Ericcson T68 cell phone does sometimes... |
| 23:37.14 | outtolunc | Duckbizkit: add a | |
| 23:37.17 | Qwell | McKillroy: Don't we all... |
| 23:37.19 | *** join/#asterisk Darwin35 (~darwin35@c-24-3-241-22.client.comcast.net) |
| 23:37.43 | outtolunc | oops |
| 23:37.46 | *** join/#asterisk scubasteve (~steve@rdu88-251-252.nc.rr.com) |
| 23:37.54 | scubasteve | Good evening!! |
| 23:37.56 | McKillroy | Qwell : I'm a bit back. I heared of * first on Saturday .... The possibility of a private GSM gate i learned friday .... |
| 23:38.02 | Duckbizkit | outtolunc you mean on the end? |
| 23:38.07 | outtolunc | Duckbizkit: whats the |3| for? |
| 23:38.07 | Delmar | <ManxPower> Delmar, I advocate renaming callprogress=yes to randomlyfuckupmycalls=yes. |
| 23:38.10 | greg_work | McKillroy, PSTN = public switched telephone network .. POTS = plain old telephone service .. you use a POTS line to connect to the PSTN (or you can use ISDN BRI/PRI .. ) |
| 23:38.11 | Delmar | I agree ManxPower |
| 23:38.19 | Duckbizkit | heh i was just changing stuff up from the other |
| 23:38.27 | Duckbizkit | the other works fine with just Queue(support) |
| 23:38.31 | McKillroy | greg_work: Thanks !! |
| 23:38.34 | Duckbizkit | this one, Queue(support) threw that error |
| 23:38.45 | *** part/#asterisk tensai (~tensai@207.141.37.66) |
| 23:38.54 | scubasteve | ooh is there a new app_fuckup_my_calls ???: :) |
| 23:39.03 | Duckbizkit | so i was adding the pipes to see if it would help |
| 23:39.07 | Qwell | scubasteve: cvs head |
| 23:39.14 | scubasteve | Qwell: LOL |
| 23:39.34 | Qwell | scubasteve: unfortunately, I added it as "app_do_something_cool", but, it sucked so bad, they renamed it |
| 23:39.44 | scubasteve | nice |
| 23:39.48 | file | woot remote MWI is working |
| 23:39.59 | Duckbizkit | outtolunc it should probably be t instead of 3, right |
| 23:40.14 | Duckbizkit | and then another pipe on the end |
| 23:40.15 | mtqh | I am having an issue with AGI, any operation that requires a user to input DTMF fails....it just skips it and moves on....any idea? |
| 23:40.18 | Duckbizkit | ? |
| 23:40.27 | outtolunc | exten => 1,1,Queue(op_ready|tn|||2) |
| 23:40.31 | outtolunc | so, yes |
| 23:40.46 | Duckbizkit | ok....well, we'll give it a shot |
| 23:41.09 | outtolunc | http://www.dynx.net/ASTERISK/DOCS/RTF/agentlogin3.rtf |
| 23:41.35 | Delmar | echotraining=800 and still self-echo on sip client. |
| 23:41.41 | fa | Can somebody send me a good extensions.conf with support for transfer of calls and voicmail accounts? |
| 23:42.25 | ManxPower | Delmar, you stopped and started Asterisk right? |
| 23:42.47 | Delmar | hell yeah. not reload. full on killed the thing and restarted. |
| 23:42.50 | McKillroy | Mother_ : Just to understand my interest: In my area I can get a UMTS phone with free internal network calls for 10 Euro a month. So - with 2 phones I could have a free mobile gate into the VoIP system. |
| 23:43.13 | *** join/#asterisk tessier_ (~treed@146.82.146.22) |
| 23:43.22 | Mother_ | McKillroy: I see, very interesting, where are you located? |
| 23:43.34 | McKillroy | Mother_ : Only 20 Euro + DSL costs would apply. I'm in Austria |
| 23:44.04 | tessier_ | Is there any way to go straight to the beep and leave a message instead of waiting through someones voicemail greeting? |
| 23:44.04 | Mother_ | very neat, so it's flat data rate for 10 euro per line? that's VERY good, here they charge at GPRS rates |
| 23:44.25 | *** join/#asterisk bjohnson (~bjohnson@ip226-181.tor.istop.com) |
| 23:44.42 | McKillroy | Mother_: Since I get a VoIP for 9 Euro I could even be called mobile for almost nothing. |
| 23:44.58 | McKillroy | I mean a callable number from POTS |
| 23:45.02 | Mother_ | lol neat |
| 23:45.08 | fa | McKillroy per motnh? 9 Euro? |
| 23:45.20 | McKillroy | A callable number , yes |
| 23:45.35 | fa | so call me ;] |
| 23:46.17 | McKillroy | Total costs for mobile telephoning in any net would be 2*10 +9 + DSL costs |
| 23:46.35 | McKillroy | with a possibility to receive calls from fixed net |
| 23:46.48 | ManxPower | + host of gateway + cost to make the damn thing work with the crappy disconnect indications most of them have. |
| 23:46.51 | McKillroy | Thats why I want that GSM gate |
| 23:46.53 | Mother_ | I can see the operator hanging you from a very tall pole :D |
| 23:47.07 | McKillroy | Rofl. |
| 23:47.40 | McKillroy | The clerk in the handy shop told me they found a couplke who used two handys as a babyphone ... because it was free .... |
| 23:47.50 | Mother_ | jeez |
| 23:47.58 | Mother_ | I bet the babies glow in the dark now |
| 23:48.06 | McKillroy | ROFL |
| 23:48.23 | fa | do anybody have standard extensions.conf? |
| 23:48.28 | fa | sample |
| 23:48.57 | Nugget | everyone here does. |
| 23:48.58 | Duckbizkit | outtolunc i made the changes, her goes nothing ;) |
| 23:49.08 | *** join/#asterisk bonbon-home (~happy@81-86-0-190.dsl.pipex.com) |
| 23:49.11 | Nugget | if you have the asterisk source you have a sample config. |
| 23:49.32 | Duckbizkit | locate extensions.conf.sample |
| 23:49.34 | Duckbizkit | heh |
| 23:49.39 | bonbon-home | has anyone else seen a problem where sip calls originated from asterisk get cut off after around 30 seconds? |
| 23:49.47 | Nugget | locate won't work for someone who just untarred it today. |
| 23:49.48 | outtolunc | eh? |
| 23:49.54 | Duckbizkit | updatedb |
| 23:49.59 | Duckbizkit | locate extensions.conf.sample |
| 23:50.00 | Duckbizkit | hehe |
| 23:50.04 | Nugget | updatedb won't work for lots of people. |
| 23:50.08 | ManxPower | I used to have simplified sample configs online, but since nobody donated to me via paypal I took down the entire site and donated it to the asteriskdocs.org project |
| 23:50.14 | Duckbizkit | i'm just kidding around Nugget |
| 23:50.28 | *** part/#asterisk jjg (tink@216.253.86.223) |
| 23:50.51 | McKillroy | Gotta run now ... Thanks a lot for your help, especially Mother_ ... BYe .. |
| 23:51.17 | greg_work | is it possible to share groups between Zap and IAX trunks? ie, say I have 4 Zap trunks and an IAX trunk .. can I have my first three outgoing calls go over Zap, then the 4th use IAX? |
| 23:51.29 | greg_work | or would I have to do an agi to do that? |
| 23:51.36 | visionv | g'night guys, you have been a big help :) |
| 23:52.03 | scubasteve | greg_work: Hm, can't seem to hear anything out of my 841...:( |
| 23:52.50 | outtolunc | as a 'group' i don't think so, but as a linked macro like TRUNKLISTXYZ = 'Zap/1&Zap/2&Zap/3&IAX2/yadda' |
| 23:52.50 | greg_work | scubasteve, i assume you have it plugged in now? :) |
| 23:52.59 | file | cooooool my mailbox status just propogated from one server to another in realtime |
| 23:53.10 | Nugget | file: how's that work? |
| 23:53.10 | scubasteve | greg_work: Yep :) |
| 23:53.14 | greg_work | outtolunc, how would you use that? |
| 23:53.22 | file | Nugget: code I wrote |
| 23:53.25 | outtolunc | they are tried in sequence |
| 23:53.26 | Nugget | spiffy |
| 23:53.35 | greg_work | scubasteve, check volume? i dunno, i didn't have issues |
| 23:53.48 | scubasteve | greg_work: Ok, am able to barely hear now.. Something is wrong... am thinking defective fone. |
| 23:53.51 | outtolunc | what you 'could' also try is to assign each to a 'local/' reference and group those |
| 23:53.54 | greg_work | outtolunc, Dial(${TRUNKLISTXYZ}) ? |
| 23:54.06 | outtolunc | ndos |
| 23:54.07 | outtolunc | er nods |
| 23:54.33 | greg_work | and it wouldn't try to call out on all of them simultaneously? |
| 23:54.37 | file | Nugget: http://www.pastebin.com/235952 |
| 23:54.54 | Nugget | that's cool. |
| 23:55.06 | outtolunc | IIRC, dev&dev&dev does in sequence |
| 23:55.07 | file | working on magical CDR records now |
| 23:55.09 | greg_work | thats another thing, how do you control or know how many calls an IAX trunk can support? |
| 23:55.31 | fa | how to add field with file name with recorded call to cdr_pgsql ? |
| 23:55.49 | JunK-Y | fa: whatcha mean? |
| 23:56.12 | fa | i want to add name of file, where the call is recorder (dumped to wav, gsm or mp3) |
| 23:56.35 | *** part/#asterisk stevekstevek (~stevekste@slim-eth0.horizonlive.net) |
| 23:57.05 | JunK-Y | why not using userfield? |
| 23:57.13 | Duckbizkit | ok outtolunc |
| 23:57.16 | Duckbizkit | i made the changes |
| 23:57.20 | Duckbizkit | and still got the error |
| 23:57.22 | Duckbizkit | http://pastebin.ca/5057 |
| 23:57.23 | greg_work | outtolunc; http://voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Dial: If you wish to specify more than one channel for the Dial command to try ? remembering that it will dial out on all of them simultaneously ? separate them with the & symbol. The channels can be different types. See Examples, below. |
| 23:57.24 | *** join/#asterisk stevekstevek (~stevekste@slim-eth0.horizonlive.net) |
| 23:57.26 | *** join/#asterisk luisgrin (~luis@209.99.227.220) |
| 23:57.28 | fa | JunK-Y how can i set the user field? |
| 23:57.39 | JunK-Y | fa: show applications like userfield |
| 23:57.41 | outtolunc | ah |
| 23:57.43 | fa | JunK-Y record or monitor is for recording calls ? |
| 23:58.28 | outtolunc | Duckbizkit: "1?voice|1:answering|1" <-- what the hell is that |
| 23:58.44 | *** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc) |
| 23:58.44 | modulus_ | does vonage do sip/iax termination? |
| 23:58.51 | Duckbizkit | just some answering machine detection |
| 23:59.39 | outtolunc | why not use the stuff in that doc? |